Re: [Asterisk-Users] ZapBarge restrictions?
Just use the authenticate app. show application authenticate Damon Estep wrote: Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be monitored. We really do not want the call center supervisor to be able to hear the CEOs calls now do we? Any solutions? ___ 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] DS3 with Asterisk
Get an M13 from adtran and split it. You could also get a Cisco AS5400 Michael Blood wrote: I have done some research on the discussions that have occured on this list about DS3s with Asterisk. It seems to be dead and I have not found any active work on the project. I know that a full DS3 may have some technical limitations with why they may not work with Asterisk but I am interested in utilizing a partial DS3. Is there anyone utilizing DS3s out there with asterisk at all and if so how are you implementing it? (Splitting? Custom Drivers? Etc..) Has anybody attempted/failed/succeeded to make a DS3 to asterisk work? Michael ___ 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] * Call Monitoring
Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ 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 http://www.voip-info.org/wiki-Example+Argus+Config ___ 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] Proxied SIP
Chris Tooley wrote: I want to install Asterisk for an organization that wants it to do some call routing for them. They have a SIP provider that is going to provide one termination and one origination account. We are going to have to route a rather large number of calls (50-100,000 concurrent), but can't find any information on how to proxy calls adaquately. ___ 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 look to SER but 100,000 calls requires a tremendious amount of bandwidth, make sure you have 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] 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
Re: [Asterisk-Users] Can Asterisk handle calls that get picked up by answering machines?
Gabriel Afana wrote: Just wondering because right now I can have it call my phone and play a message, but if I dont answer it eventually goes to voice mail. It always leaves a voicemail and when I listen to it its always the last few seconds of my message that I had Asterisk play. How do I get Asterisk to pause and wait to playback *IF* its an answering machine or voicemail? Gabe ___ 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 You can hack the record application to wait for silence. -- Todd Lieberman mailto:[EMAIL PROTECTED] http://tlsolutions.net 215.495.0030 (p) 215.495.0031 (f) ___ 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] Dialing out to 2 clients simultaneously
This is not possible... Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same peer and want them all to ring at the same time without having to set up different usernames and passwords for all these ua's and having to make difficult dialplans Is this possible? Am I doing something wrong or is this behaviour by design? Regards, Niels ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transferring variables with IAX2?
Roy Sigurd Karlsbakk wrote: is it, or can it be possible to transfer stuff like HANGUPCAUSE or RDNIS over IAX2? This is really a nessicity for multi-server setups to become any good... There is a patch floating around (on the mailing list and/or on the bug tracker) that transports the HANGUPCAUSE over IAX2 in a text message. Perhaps this could be generalized to allow any user defined variable to be passed? I beleive I read some discussion on the topic, and compared to the php register_globals case. we'll probably need some way to distinguish an external variable (sent via IAX2) and an internal variable (global or not). roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users A bad hack be to use the URL option in the Dial command. Does this idea suck? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astcc needs AGI.pm...where is it?
Bruce Komito wrote: Greetings, I tried to build astcc, but the Makefile is looking for Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be and where it comes from? I've dragged in everything I can think of from cvs, and * is otherwise running fine. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It is looking for http://asterisk.gnuinter.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phone
Cian O'Sullivan wrote: Hello, I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? Cheers Cian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Snom 220 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxy to iaxy call drops out of show channels
Sure, the IAXy's do a reinvite and * drops out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Saturday, December 04, 2004 10:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iaxy to iaxy call drops out of show channels I place a call from an IAXY to an IAXY device. INitially the calls show in the output of show channels. Then after a few seconds the show channels command shows 0 active channels even though I am still talking on the channels. Any ideas on this? THanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxy not hear ringing
Make sure you put a DigitTimeout use the r option in the Dial command. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Friday, December 03, 2004 5:01 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iaxy not hear ringing I have a couple Iaxy's and when calling out on them I dont hear ringing. Everything else is working fine. Any ideas? THanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to make/recieve call using asterisk whenthereis a power failure?
Have a look at http://www.twacomm.com/Catalog/Model_PF-6A.htm As for T1/E1, you have a big business, get a decent UPS and a generator. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Duane CoxSent: Thursday, November 25, 2004 10:18 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to make/recieve call using asterisk whenthereis a power failure? We use several Dell 2650 servers. Order them with the dual DC power supply option. Buy a row of -48 batteries and a -48 power source, your servers will stay up for hours. - Original Message - From: TinKoon To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 25, 2004 3:56 AM Subject: [Asterisk-Users] How to make/recieve call using asterisk when thereis a power failure? Hi, I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. In some situations, even though you may have a ups for the Asterisk, network equipment, channel banks, etc, but your ATA, IP phones which located near to your users and probably not connected to the UPS, so these devices wont be able to function. And even if you have a ups, after an hour or two, your uos will drain out, so how? Though we can have few analog pstn lines as standby, but these lines are mostly use for making outgoing calls rather than receiving incoming calls. For a prolong power failure situation, these lines cant really help much, so businesses will be seriously affected. It is possible to contact the telco to re-direct the incoming calls to the standby analog lines, however, it will generally take couple of hours for the telco to make the switch and very likely there will be a fee involve. I read from this forum that many asterisk implementations had been carried out, I wonder how these implementation take care of the power failure situation? Can someone share the views and implementations? ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to decrease the speech volume for record?
use sox. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Guild Jackson Sent: Wednesday, November 24, 2004 12:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to decrease the speech volume for record? Hi there, I would like to know if it is possible to decrease the speech volume that asterisk aplies into the wave form transmitted. Actually I4ve been running some tests with the purpouse to transmitt a speech from asterisk A to asterisk B and, then, in asterisk B, record this speech into a .wav file. I have noticed that this speech is recorded with its gain anplified and it is affecting the measures I want to do. I have tried to modify the txgain and rxgain variables found in the zapata.conf files but none of these seem to affect or modify the amplification I get in the speech recorded. Can someone help me with that? Thanks in advance and best regards Guild Jackson __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line load balancing
Paul, your current method of load balancing is quite fine. Why do you want to round robin load balance? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Hales Sent: Monday, November 22, 2004 6:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Line load balancing We have 4 Telular mobile phone gateways hooked up to an Asterisk box, and we need the Asterisk box to balance the load across the 4 lines. Currently, Asterisk uses the first line, and subsequent lines when the first is busy. This means that our first line is in use almost 100%, and the last line is never used. Has anyone done anything similar to this before? We have looked at using mysql and getting asterisk to query mysql to choose which line to use, but this seems overly complex. Does Asterisk have a built in function to do this simply and easily? Regards, PaulH East Hawthorn Australia CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call failover and redundancy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig
Waddington
Sent: Wednesday, November 10, 2004 5:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call failover and redundancy
Recently our provider had an issue, so we couldn't make VOIP calls.
We currently have a BRI which we use for incoming calls, at the moment I
have the below in my dialplan, so if our VOIP provider or our internet
drops, the outgoing calls are sent through the ISDN Bri.
The problem is, it takes 30 seconds of trying the IAX account, before it
uses the BRI, is there a timeout I can insert somehow, so if a call fails on
VOIP, a few seconds later it switches to the ISDN outgoing?
My current Extensions.conf
exten = _[68]X,1,Dial(IAX2/user:[EMAIL PROTECTED]/44${EXTEN})
exten = _[68]X,2,Dial(${ISDN1}:${EXTEN})
exten = _[68]X,102,Congestion
exten = _[68]X,103,Busy
--
This is how I do it. This way if you get a busy signal from the first
server, you don't dial out on your ISDN line. However, if server 1 is down
you'll go out over your ISDN line.
exten = _.,1,Dial(IAX2/server1/${EXTEN})
exten = _.,3,Dial(IAX2/server2/${EXTEN})
exten = _.,102,GotoIf( $[ ${DIALSTATUS} = BUSY ] ? 110 : 3 )
exten = _.,104,GotoIf( $[ ${DIALSTATUS} = BUSY ] ? 110 : 105 )
exten = _.,105,Congestion
exten = _.,110,Busy
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RE: [Asterisk-Users] Modifying CDR data?
If you are in AGI... make your own call log. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Saturday, October 30, 2004 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Modifying CDR data? Roy, On Sat, 2004-10-30 at 15:31, Roy Sigurd Karlsbakk wrote: I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this? I don't think you can change dst from the extension flow just like that (maybe via an app, but that might have alternate consequences) I've done some scripting with entirely different purposes, but it may fit your needs: create an AGI script that is called when a call comes in, use that to store the uniqueid of the call leg into a database. Then check if call diversion is active and log that too. Afterwards, check (i.e. once an hour or whatever is convenient) and match CDR versus your own database. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy Call Waiting Disable
Leonardo Gomes Figueira wrote: Hi, anyone knows how to disable call waiting on IAXy for every call ? I know that *70 disable for the current call but for each call I have todial it again. On dialplan I can use CheckGroup to limit the number of calls but on Queue with strategy RINGALL new calls keep ringing on the IAXy and the call waiting beep it's pretty noisy. Thanks, Leonardo I tried putting callwaiting=no in iax.conf but no help there. Any other suggestions folks? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iLBC/PCM16 Huge Cost
Wo trevor, Format and start over? Don't go crazy, just remove the files created by make install. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Trevor Peirce Sent: Saturday, October 23, 2004 10:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iLBC/PCM16 Huge Cost Brian West wrote: REMOVE THAT POS and install mpg123 0.59r, compile from src. Done and done. FYI you may want to update http://voip-info.org/wiki-Asterisk+mpg123+redhat as that's where I got inspired to download the RPM. I just stopped asterisk and killed off all the mpg123 processes... ran safe_asterisk and it immediately spawned three mpg123's (which are 0.59r). I don't see them eating up any processer time just yet but it seems to take a few hours for that to happen. I will report back later. Probably related to whatever is causing my other headaches - MOH sounds very staticy. The time, pitch, speed are all fine, but there are lots of scratch sounds and glitches added. This is with both my own MP3s and the ones included with *. I'm starting to think a format and reinstall might be a good idea there has got to be something deeper to this. Trevor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI comand channel status]
Look to hack the record app and listen for silence. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Thursday, October 21, 2004 10:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AGI comand channel status] Sir, Thanks for the reply. I have tried the callprogress in zapata and it was not reliable for me. I have tried the backgrounddetect and it does not give me busy indications. Dialogic had a callprocess function that gave busy status, waited for voice energy (some one saying hello) so you new when the phone was answered etc... I am not finding an all encompasing similar function in asterisk yet. Any further suggestions are welcome. Jerry --- Hello, On Thu, 21 Oct 2004 10:43:01 -0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / I am attempting to place a call in the outgoing spool directory // once that call is placed run my agi. I am using analog cards. / The problem is that there is no reliable call progress on ANALOG lines. Asterisk just set the channel as ANSWERED or UP as soon as a call is placed. You can try enabling callprogress in /etc/asterisk/zapata.conf , but you might have unreliable progress detection and many false hangups. The comments on zapata.conf state: ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. Also, it is ONLY configured for ; standard U.S. tones You might also want to take a look at application BackgroundDetect, just issue: show application BackgroundDetect from the CLI Good luck, -- Nicolás Gudiño Buenos Aires - Argentina -- Jerry Geis MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 (240)282-0319 Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Balaning on 2 E100P cards
I would like to know if it's possible to load balance calls on 2 E100P cards? In fact, I had an asterisk with a TE410P. 2 E1 are connected to the operator, and 2 others to an IVR PBX. Asterisk is used to place some calls in Voice over IP. I would like to know if it's possible, when I receive a call from my operator, if I can load balance it on my 2 others E1 connected to the PABX. I this case, if one PABX fail, I still had another one. show application congestion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
Florin Andrei wrote: I need a cheap simple adaptor for analog phones to use with Asterisk. It should be some kind of configure and forget type of device, to use at the office, or just throw it in a road warrior's bag and use it while travelling, to call back to the mothership. I can't decide between iaxy and sipura. Can you guys help? Which one would you use? (and why?) I feel that iaxy might have an advantage while piercing through NAT firewalls (at hotels and such), because of IAX, but i could be wrong. Or can you recommend something else? For configure and forget, I would not leave home w/out my IAXy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 and Asterisk Woes
Dan Mahoney, System Admin wrote: Okay, this one is driving me nuts. I have a fedora core 1 machine running asterisk from CVS. Built last week. I have a couple of snom phones with the latest firmware. Here's the issue, it's a wierd one. You start up the phones, they register, all is good. They show up in sip show peers like thus: danm/danm65.125.237.91D N 255.255.255.255 5060 OK (29 ms) We pass a few calls in and out, and asterisk deadlocks (not a true deadlock, see below). The sip show peers list becomes frozen. One of two things will happen: 1) I can power down the phone and it will still show status OKAY. 2) Or, the other thing I'm seeing is that the phones will forget to re-register. As in, they show up in sip show peers as status UNKNOWN, but under this non-deadlock'ed deadlock, they can still make outbound calls fine. Does anyone have any idea what can cause this? -Dan Mahoney Looks like a firewall issue too me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ResponseTimeout, Straight to operator?
Try This... exten = s,1,Wait,2 exten = s,2,Answer exten = s,3,DigitTimeout,3 exten = s,4,ResponseTimeout,10 exten = s,5,Background(ts_welcome_en) exten = s,6,Dial(Zap/3,20) Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashing with no indication why.
TC wrote: I'm hoping someone might have seen this before because I'm just about at a loss of what to do. I have an asterisk system setup in a call center environment with multiple queues. After a random uptime asterisk will suddenly come to a partial halt where I can connect to the cli but issuing a command such as show channels gives no response, and calls cannot be made in or out. Calls in progress usually drop as well, but if they don't right away, after a minute or so they will. To remedy the problem I have to do a restart on asterisk, which of course makes all the agents have to login again and is just a big mess. I have agents being dynamically added to the queues via an AGI script, also the agents are added to all queues so that they can take calls from any of them. I'm not sure if this is important but since I use the AgentCallbackLogin function I have all the agents inside their own context so that I can use a macro to determine if they are on an outgoing call (using app_checkgroup) before ringing them to prevent call waiting tones. I've thoroughly searched the messages log, in which I have both verbose and debug logging enabled. I've never found anything to indicate a problem, it simply looks like calls just slow down and stop. One other thing that may be important, I have a daemon running which stays connected to the manager api listening for events and sending off two commands every 10 seconds, one to get the status of the queues, and one to get the status of agents. say hello to app_noQ and chan_deadlock :) http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging I might be wrong about chan_deadlock but i am pretty sure yiou have dead lock situation might be cause by manager blocking all else... try the dead lock debug to see if tou can id when threads are competing for the same locks I'm with TC. I'm sure you have a deadlock. Same thing happened to me with IAX and fax detection. I turned off fax detection and my problems were solved. TL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Service Agent Auto Login
Title: Message Look at the 7905G phone from Cisco. TL -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael Blood, Matraex, Inc.Sent: Wednesday, June 30, 2004 2:50 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Answering Service Agent Auto Login Hello all, I am building a software based on asterisk to handle incoming answering service calls. I have one problem that I have not been able to figure out a reasonably pricedsolution to: The goal of this software is to allow the agent to be able to do their entire job from the desktop. The only thing that seems to be a problem is getting the operator (agents) headset logged on to the asterisk system using a computer command. (Meaning we don't want them to have to touch the phone or a headset). We have done most everything else we need through the Manager API but this onehas us stumped. We need to use IP Phones or some sort of IP based extensions for flexibility. Here are a few things that we have conceptually tried. Auto Answer: We set up an IP Phone with Auto Answer turned on. Then when the operator says that they will accept a call we route the call through to their phone and Auto Answer picks it up. Sounds like a great fix but the only phones we can find with Auto Answer are more expensive with lots of other features that will never be touched. (Cisco 7940 ...) In fact we would not even want the actual phone to be visible or usable to the operator. It would be hidden or locked in a desk drawer with the head set cord coming out of it. So... a phone with auto answer COULD work if we could find an inexpensive enough one (less than $150 would be okay) any suggestions would be great. Agent Queue: We setup an Agent Queue that the agent has to dial into at the beginning of their shift. The problem here again is that we do not want the agent to have to touch the phone itself. The agent Queue COULD work if we found a phone that we can program to automatically dial in to the queue each time that the line was picked up. Then we could put some sort of headset on the phone which has an on off switch that allows the agent to connect or disconnect the phone from the server. I just don't like the fact that the operator would have to do both of those things. I suppose the computer could prompt them to make sure they turn on their headset and that would work if there was no other solution. Does anyone know of a solution where I would be able to setup some sort of permanent connection to the asterisk server via IP? I can't have a dial tone in their ears constantly and I need to find a phone or solution which is $150 or less (preferably under $100) per workstation. How are existing answering services dealing with this problem? (Maybe they don't use IP?) Thanks for any help or direction you can give me. Michael Blood PS. I have found that the Grandstream Budgetone has auto answer on it but that it wont support a headset for another 2 - 3 months (more likely 6-9) and I am looking for a solution which will be ready in about 1 month.
RE: [Asterisk-Users] Asterisk and dial-up modems
Title: Asterisk and dial-up modems Look at the ZapRAS 'show application ZapRAS' this only work w/a PRI. TL -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of John VogelSent: Tuesday, June 29, 2004 11:24 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and dial-up modems Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other?
RE: [Asterisk-Users] SFTP
Your WSFTP program may only have SSH1 but your Debian server may only have SSH2. Look in /etc/ssh/sshd_config Make sure you have 'Protocol 1' I do not recommend this setting as it is not secure. I use F-Secure SSH Client w/Debian and like it. TL P.S. Please take this question to a debian or wsftp support list if this suggestion doesnot solve your problem. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: Thursday, June 17, 2004 7:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
RE: [Asterisk-Users] IAX registration
Sounds like a firewall issue to me. How does your FW handle state. TL -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of SathyaSent: Wednesday, June 16, 2004 2:23 PMTo: [EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] IAX registration Hi, I have a nufone connection (IAX2), works fine. In my iax.conf I do not specify a time interval that * needs to renew registrations with nufone server. However I can see following registration messages on my cli every 90 seconds (approximately) --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569--Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569--Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569 Does IAX2 renew registration by default every 90 seconds ? Since my server is always connected, wouldn't there is a possibility to set registration interval to a higher value. I tend to think that nufone server is loosing my * and hence my callers get "number unreachable message" due to the fact that I have to keep on registering with it. Any help appreciated. Cheers Sathya
RE: [Asterisk-Users] Queue then Voicemail
; goto philly q exten = 0,1,Answer exten = 0,2,Background(wrn-phillyq) exten = 0,3,Queue,phillyq exten = 0,4,WaitMusicOnHold(90) exten = 0,5,Voicemail(u1) exten = 0,6,Playback(vm-goodbye) exten = 0,7,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Sent: Tuesday, June 15, 2004 4:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Queue then Voicemail Hi all, I'm stuggling with how to present calleds to a specific DDI (DID) with Music on hold whilst the call is hunted around 3 phones, then if not answered within a certain period forwarded to voicemail. So far I've got the queue working and the voicemail but not both together. Ive had a look on the wiki and the archives but can't spot anything that might point me in the right direction. CHeers Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Ditto on Avaya... My $75,000 Avaya Definity G3Si has a GUI that simply wrapps the CLI. If you don't understand the CLI you can't use the GUI. Their Java apps for their interaction center / ip office suck, I prefer the .conf solution. Easier version control and more concrete. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Colin Anderson Sent: Wednesday, June 09, 2004 11:51 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony I like the way the 3com NBX system works. The web interface is pretty intuitive. Adding users and devices is a snap through the GUI but to get to the real meat you have to edit the dial plan. To do this, you download a text file to your desktop, edit it, then upload it again. Ditto on the Mitel ICP 3300. It's just a GUI layer on top of their command line crap that they dusted off from the SX-2000. Mitel had a great opportunity to redefine PBX managment and they kind of p*ssed it away because their managment stuff was designed by engineers, not GUI designers. At this stage, from what I can see, there's no functional difference between configuring * vs my 3300. So, take heart, * users, Mr. Spenser's little project is, IMHO, equivalent to what an army of Mitel engineers took years to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp wont compile.
add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS login
Install CVS. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Fabio Donaggio Sent: Thursday, May 27, 2004 11:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CVS login Hi to all!! Here is my problem: [EMAIL PROTECTED] root]# cd /usr/src [EMAIL PROTECTED] src]# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot [EMAIL PROTECTED] src]# cvs login -bash: cvs: command not found [EMAIL PROTECTED] src]# Anyone can help me?? Thanks for all!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zhone Zplex issues
Me too... Sometimes ports go bad on the Zhone... You just can't use the ports that go bad. Time to get a new zhone. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kent Williams Sent: Tuesday, May 25, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zhone Zplex issues Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from throwing the Zhone out)? Cheers, -Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 NAT / Registration Issue
His firewall is stateless. I've run into the same issue w/the sonic wall firewall on a client site. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol Sent: Sunday, May 23, 2004 11:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX2 NAT / Registration Issue I have a client using IAX Phone at his office to connect to his Asterisk located at a data center. His IAX Phone connects through his office NAT gateway device (unfortunately I don't know the specific brand and model). He can make calls just fine. However, he seems to have issues receiving calls. I monitored his Asterisk for a bit and noticed that his IAX Phone, which registers every 60 seconds, always registers from a different port. Does this indicate that 60 seconds is too long, and that the NAT is closing the hole between registrations? I will try to find out what type of router/gateway he is using and post it. Any thoughts would be appreciated. Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA devices
Multitech make an 8 port SIP device. Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA devices
Check out http://www.mediatrix.com/documents/datasheets/Mediatrix_1124_0302v0.pdf Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on OS X
Abandon OSX. Make your system dual boot with Gentoo or Yellowdog and compile on Linux. It's just easier. If you have an old PII I'd use that instead. TL -- Todd Lieberman [EMAIL PROTECTED] http://tlsolutions.net p. 215-495-0030 f. 215-495-0031 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ed Mansouri Sent: Tuesday, May 18, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk on OS X Hello, I have researched a few postings where users mentioned being able to install Asterisk on Mac OS X Panther by adding some code after line 165 in the Makefile and then compiling. This has been unsuccessful for me. I downloaded the asterisk-0.9.0.tar.gz tarball and am trying to install from it. The output I get upon trying to make after editing the Makefile can be viewed at: http://support.ucompass.com/make.txt If anyone has any experience getting Asterisk to work in Mac OS X and has any suggestions for me, I'd appreciate it. Ed - Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book http://support.ucompass.com/addressbook.html Committed to Building Profitable E-Learning Enterprises Phone: (850) 297 1800 x 201 FAX: (850) 553-9252 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 N:Lieberman;Todd;E;Mr. FN:Todd E Lieberman ORG:TLSolutions, Inc TEL;WORK;VOICE:215.495.0030 TEL;WORK;FAX:215.495.0031 ADR;WORK;ENCODING=QUOTED-PRINTABLE:;;P.O. Box 554=0D=0A;Lafayette Hill;PA;19444 LABEL;WORK;ENCODING=QUOTED-PRINTABLE:P.O. Box 554=0D=0A=0D=0ALafayette Hill, PA 19444 EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20040411T154025Z END:VCARD
RE: [Asterisk-Users] Dropped calls
do a 'sip debug' and make sure all looks good. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito Sent: Monday, May 17, 2004 10:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dropped calls I'm having a problem with outgoing dropped calls. They symptom is, when I place a call from a sip extension to the outside, the call is connected properly, but then abruptly disconnects anywhere from 10 to 60 seconds later. This happens when the outgoing call is through a POTS line (TDM) as well as over a sip gateway. Calls between sip extensions do not have this problem. Has anyone ever experienced this? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
You'll also have to modprobe the x100p /sbin/modprobe -k wcfxo /sbin/modprobe -k zaptel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Sunday, May 16, 2004 9:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] (no subject) Hello I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card and one USB one port FXS card. I can modprobe wcusb but ztcfg always return ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message. Also unable to config outgoing call using SIP SoftPhone. Any working examples of configuration files is highly appreciated. I mentoned followin lines in /etc/zaptel.conf file. fxsks=1 fxoks=2 Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
What does 'cat /proc/interrupts' tell you? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Deepak Malhotra Sent: Sunday, May 16, 2004 11:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] (no subject) i did that bit no luck. - Original Message - From: Todd Lieberman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 16, 2004 6:47 PM Subject: RE: [Asterisk-Users] (no subject) You'll also have to modprobe the x100p /sbin/modprobe -k wcfxo /sbin/modprobe -k zaptel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Sunday, May 16, 2004 9:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] (no subject) Hello I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card and one USB one port FXS card. I can modprobe wcusb but ztcfg always return ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message. Also unable to config outgoing call using SIP SoftPhone. Any working examples of configuration files is highly appreciated. I mentoned followin lines in /etc/zaptel.conf file. fxsks=1 fxoks=2 Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] Concept for line appearances and bridging: anyone?
John, i think MGCP has this feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Friday, May 07, 2004 5:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Concept for line appearances and bridging: anyone? OK, here's a configuration challenge: I want to have certain line appearances able to be interrupted by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all ring on line appearance 1234 (there is a specific line labelled 1234 on each phone) 3) User A picks up the ringing call on 1234. Line X and User A are bridged. 4) User B saw the caller ID on the call before it was picked up by user A, but she wants to talk to the caller as well since she has some relevant information. User B picks up the phone and pushes the 1234 extension button. A warning tone is played into the conversation between X and User A, and then User B is bridged into the conversation. User B then talks with X and User A, and then hangs up. This is _extremely_ relevant to office PBX systems. In fact, it's one of the most used features - the ability to share a call with other people in the office just by hitting the right line appearance button. Has anyone come up with a reasonable solution to delivering this feature? For small offices, this is really a mandatory feature though as the number of calls increases this becomes more useless in an inbound setting (though as a workgroup feature it gains usefulness with size of the organization. I'll skip the business cases for why this is a good idea and leave it as an exercise for the reader.) I have come up with ideas on doing this with some really horrible, nasty, awful ideas that involve MeetMe rooms, but shudder... they're really not the right way to do it. There must be some clever way of doing this with a new channel specification that would allow bridging into an existing channel identifier. I.E.: Dial(Bridge/SIP/2203-bed5) Other related topics: - The auto-dial I can handle with PLAR (hotline calling - pick up the phone, and automatically a number is dialed) and DISA on the Asterisk side. In other words, when someone picks up line #1 on their Cisco 7960 (or whatever phone) I can have the system auto-dial into my * server. Using the caller ID, I can determine what line they're calling from. If there is nobody on that line appearance, then I can give them a DISA to allow them to dial a regular call, as if the auto-ringdown didn't happen. - This feature becomes useful now that we have some phones that support SUBSCRIBE methods to allow other phones to show who is on what lines. We can _see_ who is on the line, but there is no ability to add other lines to the call without transferring to a MeetMe (which then causes call control to be lost, and is a hassle, etc. etc. etc.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load module chan_zap.so failed
you must ztcfg -vv then modprobe your zaptel harware... jorge verastegui wrote: Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L -- *NOTA DE REDCETUS S.R.L.* : La información contenida en este E-mail y sus anexos, sólo puede ser utilizada por el individuo o la compañia a la cual está dirigido. Si no es el receptor autorizado, cualquier retención, difusión, distribución o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo -- Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E100P for Bandwidth Termination
If memory serves correct, you'll you have to pass the data channels to an open T1 card and use a cross over cable to a router. Don't quote me on this as I have not done this myself. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Azher Amin Sent: Saturday, April 17, 2004 1:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E100P for Bandwidth Termination Hi, I have a query from a client that can he use the E100P card to terminate the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco router ?? The other end is a cisco 2620 router with E1 VWIC-1MFC. Can anyone explain if its possible with Asterisk and further any configuration help. Applreciated. Regards Azher Amin --- http://www.consulttech.com.pk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] libspandsp.so.0
copy libspandsp.so.0 to /usr/lib/asterisk/modules -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 18, 2004 7:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] libspandsp.so.0 I successfully compiled installed the spandsp-0.0.1k.tar.gz modules for faxing and patched the asterisk according to the readme and rebuilt and installed * but I am getting this error when attempting to start *. The libspandsp.so.0 file exists and I have coppied it to several directories recompiled and have the same results. What am I doing wrong? help please [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: loader.c:239 ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory Apr 18 18:57:20 WARNING[1024]: loader.c:407 load_modules: Loading module app_rxfax.so failed! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] no sound when connected
Sounds more like a firewall issue. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Saturday, April 17, 2004 6:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] no sound when connected I'm having a sound issue. I'm using BT100 (102). When I dial the echo test ( or anything for that matter) outside of my LAN there's no sound when it answers although I hear the ringing tones. Is this an RTP or codec issue. When I dial through Zap everything is fine. Thanx. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Run Asterisk without any .conf file ??
after 'make install' run 'make samples' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Angel Diaz Sent: Wednesday, April 14, 2004 11:00 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Run Asterisk without any .conf file ?? Hi all, I am very new with Astersik. Could some body tell me if it is possible to run Asterisk without any .conf file in /etc/asterisk ? I just want to test if my Asterisk has been installed correctly and as I am waiting for digium cards ... I have already tried but nothing happened after some verbose it stop... Thanks Angel __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Question
du -sh /var/spool/asterisk/vm/* -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Paul TyremanSent: Sunday, April 11, 2004 11:17 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Voicemail Question Hi, Is it possible to stop users from having the option to change their voicemail greetings ? I'm running asterisk on an old machine with a small hard drive and I don't want users to be able to take up room on it by recording their own messages. Also, once someone has recorded their own messages, cant it be set back to the default one ? Once final question, is it possible to see how much space each users mailbox is taking up and can I delete voicemails from users mailboxeson the server ? Thanks, Paul.
RE: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device
modprobe your zaptel hardare... TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anon Sent: Sunday, April 11, 2004 7:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: == Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = [ Booting..Apr 11 17:42:50 WARNING[16384]: chan_zap.c:684 zt_open: Unable to specify channel 2: No such device Apr 11 17:42:50 ERROR[16384]: chan_zap.c:5357 mkintf: Unable to open channel 2: No such device here = 0, tmp-channel = 2, channel = 2 Apr 11 17:42:50 ERROR[16384]: chan_zap.c:7482 setup_zap: Unable to register channel '2' Apr 11 17:42:50 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 -- Unregistered channel 2 Apr 11 17:42:50 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! My zaptel.conf: == loadzone=us defaultzone=us # Load FXO device (a X100P PSTN card) as Channel 1 fxsls=1 # # Load FXS device (a TDM400P) as Channel 2 fxols=2 My zapata.conf: == [channels] language=en context=pstn signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=+0 txgain=+0 channel = 1 ;Channel 1 will be the X100P PSTN card language=en context=CustomerSide signalling=fxo_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 2 ;Channel 2 will be the TDM400P card Thanks in advance for any help, Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Softphone (with USB headset) for Mac recommendations
I don't have a Mac so I can't suggest a good client, but, I suggest a Cisco 7905G w/a plantronics head set, it's perfect for this job. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Tuesday, April 06, 2004 11:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Softphone (with USB headset) for Mac recommendations Well, I have my 7940 up and running at home, and everything's working pretty much perfectly. So, it's time to find new things to break :-). The problem de jour is long-distance calling from work: I can use my cell phone, but I'm in an inside office on the 42nd floor and it's really hard to get good reception. So, I'm thinking about running a softphone on my Mac laptop and using it. The big problem is that my office is *noisy*--I'm currently sharing it with a pile of Cisco routers and a number of 1U servers with beefy fans. So, I need a good USB headset that can cope with noisy conditions. Oh, yeah, and it'd be nice if it was cheap. Does anyone have any suggestions? Would I be better off spending the $65 for a cheezy budgetone then $20 on a headset, and $20 on software, and $25 on a better headset, and so forth? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
I'm keen on motv. I'll build a cgi wrapper for the logs if you need to assign the task to someone. TL On Tue, 2004-04-06 at 23:31, Mark Spencer wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time 32-bit unsigned int version which is incremented whenever some major new bug is fixed. When Asterisk starts up (and periodically, maybe once per day), it sends a packet with the version number to a server at Digium, along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium server replies (if it receives the packet, if not, it might get sent again in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are associated with that version of the code. In this way, an asterisk administrator could easily see if there were any major issues, critical security updates, etc, that his system might need to be updated for. Now, of course, any time you put a call home feature in, there are people who will be concerned about privacy. Clearly it will be able to be disabled, but I want to run my idea about deployment by everyone here and see if you guys had some ideas. The idea would be that *new* installs (make samples) would have the feature turned on for MAJOR level by default, and that any existing install (e.g. /etc/asterisk/sip.conf exists, but not /etc/asterisk/motv.conf) would have the file created at the next make install based upon prompting the installer. Any feedback on: a) The idea itself -- is it a good one or is it stupid? b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Errors
Can anyone decipher these error messages? Mar 18 18:10:21 WARNING[131081]: chan_zap.c:5949 zt_pri_error: PRI: Read on 39 failed: Unknown error 500 Mar 18 18:10:21 NOTICE[131081]: chan_zap.c:6664 pri_dchannel: PRI got event: 6 on span 1 Thanks, TL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone with large display
I use the 7905G with SIP. Only problem is there is no mic for the speakerphone. On Tue, 2004-03-09 at 20:34, Jonathan Moore wrote: I believe these are skinny only models. Have you had luck with them in an * environment? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VOIP Analog adapter ???
I like putting a TxxxP in your * system and connecting the systems via a T1 cross over cable. Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.500.6913 f. 208.485.7850 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to route based on DNIS
Alas I feel the same way, but my service provider has the 800 numbers connect to the POP via a Lucent Switch and then pass the 800 calls to a local inter connect via some other switch. Ask your provider if they can pass you the actual DNIS. In my case Verizon (Philadelphia) can but my lower rates are with a CLEC (ATX) and they can not. I only pay $3 for 20 DID's so the added cost is marginal but the management is a bit tricky as my DB schema did not consider masking local numbers to 800 numbers. You'll just have to play it by ear. Regards, Todd Lieberman On Wed, 2004-02-25 at 22:36, John Brown (CV) wrote: Hi List, how does one route calls in extensions.conf via DNIS ?? I need to route the 800 number that was dialed to the right part inside of asterisk. I don't want to waste a PSTN DID for each Watts number. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialing
Dear Matt, I have never used shady dial but would be happy to try and get it working over IAX for you. Here is how I see this working; during you initial R D phase I would put you on one of my development servers with a PRI T1 that I use for testing. Your cost during the RD phase will be $100/month for the server/bandwidth and .04/min for calls made within the continental US. When you outgrow the solution above we'll get you your own server and a dedicated PRI T1 line(s). I can offer you the dell sever below for $250/month. A single PRI T1 will cost you $300/month plus .03/min. 1) PowerEdge 1650 2x1.4GHz P3,1GB,2x18 SCSI,RAID,Dual Power 2) Digium T100P w/your own Dedicated PRI T1 line 3) Network Monitoring, Cisco PIX/35xx Switches BGP routing. 4) 8x5 Telephone support 24 hour emergency pager Thank you in advance for the opportunity to offer you this solution. Regards, Todd Lieberman -- Todd Lieberman TLSolutions, Inc. - Your Total Lead Solution 215.500.6913 mailto:[EMAIL PROTECTED] http://www.tlsolutions.net On Thu, 2004-02-19 at 21:20, matt wrote: Is there a IAX provider who can offer predictive dialing (eg shady dial) in the US and pass calls which are answered back to me here in New Zealand? Kind regards, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zhone + call transfer
Kent Williams wrote: After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If your phone has a setting for flash button timing try increasing it. I'm not sure how to tweek the timing of flash within *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Force SIP Phones to Register
Tom Green wrote: Hi, Is it possible to force a SIP phone to send a register message to the PBX? I want to change a phone's extension. By forcing that phone to send a register msg, I can ensure that the phone is able to make or receive calls without any delay. Any pointers/help is appreciated. TG __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just set the refresh rate low, say 3600 ms, or reboot the phone. TL -- Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] 215.500.6913 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My remote 15 seat call center uses 79xx phones and a point to point T1. Your millage may vary with the number of users/applications your bandwidth supports. You may need to install QoS for your network to give SIP traffic top priority. It's best to have a low latency connection! Regards, TL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Attendent
I made my own MP3 files for the hold music. TL Shad Mortazavi wrote: Dear All, I have a number of call queues defined in Asterisk. I would like to program a system attendant that tells people; 1. Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly appreciated. Kind Regards Shad Mortazavi US Technical Manager Nexus Management ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960, Nortel MICS, Digital sets, ...
Stay away from the MICS. I realize the handsets are nice but the entire system is far too complex for it's own good. The M in MICS stands for modular. My MICS supports a 15 seat call center. The system has 4 modules daisy chained together, a remote access box, a separate system for music on hold, wiring galore and a hooky user interface to program with. Worst part is if one module goes down it can bring the entire sysetm down!. The kicker for me is that I don't have someone one site so it can take hours to troubleshoot and fix my problems remotely. The MICS new cost me almost 20 G's. In contrast, I am very satisfied with asterisk. I have a 12 seat call center and the total cost was about $6000. It took me about two weeks to get comfortable with * but now I'm addicted. I suggest you take * for a test drive. Get two Cisco phones off ebay and install asterisk on any old P3 or better. Sign up for nufone.net and find out for yourself how good VOIP sounds. All my phones are Cisco 7940's. Programming the services button is easy if you can do web programming and it's reliability is fair to good or no worse than Nortel. I have no experience with paging but I hear the Snom phones can do an auto answer so if that is a requirement you may want to check the list archives for some paging background before you buy cisco, you may want Snom. Good luck, Todd Lieberman -- Todd Lieberman [EMAIL PROTECTED] tlsolutions.net 215-500-6913 Derek Billingsley wrote: I have a couple of questions I'm hoping folks can help me out with. When I search the mailing list, I see folks doing what I'm interested in so here's hoping ! - How are people making out with interfacing to the 7960? I'm considering buying a number of these as they look quite feature rich. But, are they easy to interface to? - Will I be able to interface with softkeys on the phone? For example, I see the phone has a 'Services' button that I'd like to program, maybe even with material that gives agents real time information (such as any real-time network information I want flashed to agents), - As an alternative to above, we are considering the Nortel MICS approach (like a low cost 6x16). Our ILEC supplies us currently with a CO-based PBX service and we've come to love the Meridian sets. As an alternative to the IP phones, I've seen refurb'd MICS systems for sale. I read in the group that the phones themselves are a proprietary Nortel signalling but I can front-end the PBX with an Asterisk setup. If I do this, is it possible to maintain the ability to: - .. select from a list of multiple lines that might be ringing? (makes sense if each trunk is fed separately from Asterisk) - .. intercom access between phones seems as though it would be through the MICS and not touch Asterisk at all - .. Can I light a message-waiting lamp? (So a VM waiting in Asterisk would light the lamp on a particular Meridian set) If it wasn't for those damn friendly Meridian sets haha... but maybe the Cisco set is a good substitute. - Just another final question, has anyone had success in running a real-time, low cost Internet-based trunk between office locations using Asterisk and going through setting up dialing plans, etc. I want to push ahead for a trial of this but I'd like to know what I should be expecting? Can I expect toll-quality voice? Any problems with full-duplex? etc This is a basic question, I know, but I'm trying to understand what my expectations should be. Thanks for the help, this project is quite exciting stuff. Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Norstar MICS
Check out: http://lists.digium.com/pipermail/asterisk-users/2002-December/006728.html David Gomillion wrote: I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the question: I wish to put * between the telco and the Nortel PBX. How do I set up dialing out a PRI from * to the Nortel and sending the 4 digits so that the legacy phone system knows which extension to route the call to? Am I thinking about this wrong? It seems to me that I need to buy a 4-port T1 card, take in the 2 PRIs on two of the ports, do the * magic, and output 2 PRIs on the other 2 ports, thereby connecting the Nortel and * and the P$TN. Has anyone actually done this? Can it work? I really don't want to lie... If I need to present the plan as replacing the entire system all at once, that's fine, but they're a lot more likely to sign off if we can do it as a phase-out instead of a forklift upgrade. Thanks, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
Put the calls to a Q and have your reciptionist login/logout of the Queue.
xten = 600,1,AddQueueMember(phillyq|SIP/${CALLERIDNUM:6})
exten = 600,2,Playback(agent-loginok)
exten = 600,3,Hangup
exten = 601,1,RemoveQueueMember(phillyq|SIP/${CALLERIDNUM:6})
exten = 601,2,Playback(agent-loggedoff)
exten = 601,3,Hangup
Sri wrote:
Hi All
This is one scenario I would like to have some help. I have searched
the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her phone?
Eg.
8am - Attendent enters office - switches OFF auto attendent. He/She
takes in all the incoming calls and answers.
12pm - out of lunch. Needs to put the system back into Auto. 1 pm -
return from lunch. Needs to switch OFF auto attendent
5 pm- Puts Auto attendent ON.
I am sure there can be a script built that should change
extensions.conf. and reloading asterisk on the attendent activating
based on a clock that kicks in 8 am, 12 pm, 1 pm and 5 pm.
I dont want this to be time restricted. the attendent should have
control.
Is there a better way ? this could be even done through the phone of
the attendent eg, like *80-1 (ON) *80 - 2 (OFF)...
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Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
Using a Q is nice because you can have another employee or group of
employees cover while the receptionist is on break.
TL
Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would like to have some help. I have
searched the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her
phone? Eg.
8am - Attendent enters office - switches OFF auto attendent.
He/She takes in all the incoming calls and answers.
12pm - out of lunch. Needs to put the system back into Auto.
1 pm - return from lunch. Needs to switch OFF auto attendent
5 pm- Puts Auto attendent ON.
I am sure there can be a script built that should change
extensions.conf. and reloading asterisk on the attendent activating
based on a clock that kicks in 8 am, 12 pm, 1 pm and 5 pm.
I dont want this to be time restricted. the attendent should have
control. Is there a better way ? this could be even done through
the phone of the attendent eg, like *80-1 (ON) *80 - 2 (OFF)...
exten = *801,1,DBPut(auto/attendant=1)
exten = *802,1,DBPut(auto/attendant=0)
exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($[${autoattendant} = 1]?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
Thank you all!
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant
created.
/O
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RE: [Asterisk-Users] Needed - Asterisk Consulting
Hi Sean, I spoke to Atool about his project and assured him AGI on asterisk was the way to go. I quoted $1000 for the programming and he may be interested, again, thanks for the lead. I'm in Philly but I'm only 2-3 hrs away from the DC market place. If you would be so kind to pass my contact information to your customer I would really appreciate it. BTW: I need a T100P. Please contact me to place the order. Regards, TL -- Todd Lieberman 215-500-6913 [EMAIL PROTECTED] http://tlsolutions.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean P. Robertson Sent: Thursday, December 04, 2003 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Needed - Asterisk Consulting A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean ___ Sean Robertson NETXUSA p. 800-289-6389 f. 864-233-4344 Ask me about Voice over IP. http://www.netxusa.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Draft RFP for Asterisk installation/configuration
Dear Rob, I'm a solo Asterisk vendor in Philadelphia, PA (www.tlsolutions.net). I would like to submit a reply to your RFP. I would also like you to consider other hardware recommendations for your * systems. Are you available by telephone to discuss your project in more detail? Regards, Todd Lieberman 215-500-6913 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Page Sent: Thursday, December 04, 2003 10:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Draft RFP for Asterisk installation/configuration Hello everyone: I've been lurking on this list for a bit now and reading about *. The project seems to have great momentum and could-always-be-better documentation. I can relate! :^) We would like to get some help from the experts on this list getting a prototype installation installed at Zope Corporation. We've posted a *draft* RFP at: - http://www.zope.com/AsteriskRFP I would like to get comments on the draft RFP (technical correctness, clarity, requirements, etc.). We'll work to incorporate these comments into the RFP and then post another note inviting interested parties to write proposals. We are located in Fredericksburg, VA (about 40 miles south of Washington, DC). That said, presuming our approach in the RFP is valid we don't need a local vendor - all work can be done remotely. I would also appreciate it if the integrators/vendors/solo gurus out there who are inclined to make a proposal would send me a short note indicating this interest. Thanks in advance, Rob P.S. - Feel free to send me email off-list. We will publish the comments we receive (removing people's identity) and document any modifications that we make to the RFP on our website. -- Rob PageV: 540.361.1710 Zope CorporationF: 703.995.0412 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pressing 0 in Voicemail causes * to hangup
The Problem: When a call gets into voicemail from Queue and presses 0 before leaving a message * will issue a Hangup. I'm sure it's a context thing I just don't know where it is. Any suggestions would be appreciated. Regards, TL -- Playing 'vm/1/unavail' (language 'en') -- Hungup 'Zap/2-1' Here is a snip-it from my extentions.conf [qout-phillyq] exten = 0,1,Voicemail2(u1) exten = 0,2,Goto(default,s,1) [open] ; goto philly q exten = 1,1,Answer exten = 1,2,Background(wrn-phillyq) exten = 1,3,Queue,phillyq exten = 1,4,WaitMusicOnHold(20) exten = 1,5,Voicemail2(u1) exten = 1,6,Playback(vm-goodbye) exten = 1,7,Hangup Here is my queues.conf [general] ; ; Global settings for call queues ; (none exist currently) [phillyq] music = default announce = queue-phillyq context = qout-phillyq timeout = 15 retry = 5 maxlen = 0 -- Todd Lieberman 215-500-6913 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup
I'm using Voicemail2. Either way my systems issues the hang up w/v1 or v2. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, November 24, 2003 12:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup Voicemail1 is gone. Voicemail2 replaced voicemail early this month. bkw On Mon, 24 Nov 2003, Tim Thompson wrote: I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten = 0,1,Voicemail(u1) exten = 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227 -Original Message- From: Todd Lieberman [mailto:[EMAIL PROTECTED] Sent: Monday, November 24, 2003 9:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup The Problem: When a call gets into voicemail from Queue and presses 0 before leaving a message * will issue a Hangup. I'm sure it's a context thing I just don't know where it is. Any suggestions would be appreciated. Regards, TL -- Playing 'vm/1/unavail' (language 'en') -- Hungup 'Zap/2-1' Here is a snip-it from my extentions.conf [qout-phillyq] exten = 0,1,Voicemail2(u1) exten = 0,2,Goto(default,s,1) [open] ; goto philly q exten = 1,1,Answer exten = 1,2,Background(wrn-phillyq) exten = 1,3,Queue,phillyq exten = 1,4,WaitMusicOnHold(20) exten = 1,5,Voicemail2(u1) exten = 1,6,Playback(vm-goodbye) exten = 1,7,Hangup Here is my queues.conf [general] ; ; Global settings for call queues ; (none exist currently) [phillyq] music = default announce = queue-phillyq context = qout-phillyq timeout = 15 retry = 5 maxlen = 0 -- Todd Lieberman 215-500-6913 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Stops Responding
My last restart was 8 days ago. I'm now running: 'asterisk -vgc /var/log/asterisk.log' from screen 0 and 'tail -f /var/log/asterisk.log' from screen 1 I have not had a crash since 8/25 and have run about 6580.18 minutes through the system over my PRI. Hardware: VALINUX 1220 PIII 866, 384Mb, 2x20 GHZ T100P cdr_mysql is being used and is 8/25 cvs. I do not load the ;driver=aopen in modem.conf Are you using the uniqueid feature from within cdr_mysql? I am, but I hacked it to work. I'm sure there is a place for me turn the uniqueid feature but I did not know where to turn it on from so I removed the if statement and force the insert of the uniquie id. Maybe that has something to do with it. Regards, TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Congdon Sent: Wednesday, September 03, 2003 8:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Stops Responding This is getting to be a big problem. I am hoping it is something I have setup wrong somewhere... Various channels just freeze. It always appears to be the agents phones only. They will come to me and say the phones are down again. This morning here is what I see. I can not do STOP NOW. Just returns to the CLI prompt. I have to kill it. Notice that I try to hangup the channels and nothing happens. Any suggestions? pbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) 7 active channel(s) pbx*CLI soft hangup Za Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI soft hangup Zap/ Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI soft hangup Zap/ Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI soft hangup Zap/66-1 Requested Hangup on channel 'Zap/66-1' pbx*CLI soft hangup Zap/65-1 Requested Hangup on channel 'Zap/65-1' pbx*CLI soft hangup Zap/64-1 Requested Hangup on channel 'Zap/64-1' pbx*CLI soft hangup Zap/54-1 Requested Hangup on channel 'Zap/54-1' pbx*CLI soft hangup Zap/52-1 Requested Hangup on channel 'Zap/52-1' pbx*CLI soft hangup Zap/25-1 Requested Hangup on channel 'Zap/25-1' pbx*CLI soft hangup Zap/26-1 Requested Hangup on channel 'Zap/26-1' pbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) 7 active channel(s) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reload not working
Martin - et all, I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco Phones w/SIP load 4.4. What hardware do u have? The worst part is that my system will sometimes just busy out even if I do not issue a reload command! However if I issue reload it's a sure thing * will hang. I spoke w/[EMAIL PROTECTED] about this last week but we never resolved it and my system has been hung 3 times in two weeks. Each time it hangs I killall -9 asterisk, cvs update restart. He said he was working on the releasing of a PRI channel code last week and that the Telco and * may not be jiving correctly. I had the problem w/the 7/8/2003 cvs so the issue probably still exists. I'll report any findings from asterisk -gc as soon as I find any on my side, I just don't want to take down * during business hours. So tonight I'll restart w/asterisk -gc try and get to the bottom of this w/you. TL Here is what he [EMAIL PROTECTED] recommended: Well if you use sip phones you might want to have sip debug turned on. But if you have many SIP phones then you're going to have lots of SIP messages. The zap show channels output is broken. It might show some false messages. Instead do zap show channel chann_no If the PRI Flag is Call then this channel hasn't been cleared properly. If it's empty than it's ok. If the channel is not serviced by your telco it's going to be in Restarting state. Also make sure then when you restart asterisk that all the 23 channels get restarted at the very begining. regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] reload not working I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reload not working
What is your SIP registration timeout? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 11:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] reload not working I have three Cisco 7960 phones/SIP 5.3 using two Wildcard X100Ps and IAX service from Nufone. It worked fine on my earlier installed CVS from 6/10. I have not noticed any random hangs, altough it has only been running for two days. Thanks, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Lieberman Sent: Wednesday, August 20, 2003 11:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] reload not working Martin - et all, I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco Phones w/SIP load 4.4. What hardware do u have? The worst part is that my system will sometimes just busy out even if I do not issue a reload command! However if I issue reload it's a sure thing * will hang. I spoke w/[EMAIL PROTECTED] about this last week but we never resolved it and my system has been hung 3 times in two weeks. Each time it hangs I killall -9 asterisk, cvs update restart. He said he was working on the releasing of a PRI channel code last week and that the Telco and * may not be jiving correctly. I had the problem w/the 7/8/2003 cvs so the issue probably still exists. I'll report any findings from asterisk -gc as soon as I find any on my side, I just don't want to take down * during business hours. So tonight I'll restart w/asterisk -gc try and get to the bottom of this w/you. TL Here is what he [EMAIL PROTECTED] recommended: Well if you use sip phones you might want to have sip debug turned on. But if you have many SIP phones then you're going to have lots of SIP messages. The zap show channels output is broken. It might show some false messages. Instead do zap show channel chann_no If the PRI Flag is Call then this channel hasn't been cleared properly. If it's empty than it's ok. If the channel is not serviced by your telco it's going to be in Restarting state. Also make sure then when you restart asterisk that all the 23 channels get restarted at the very begining. regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] reload not working I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Killing runaway PBX
kill -9 PID -- Todd Lieberman 800-675-3078 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Friedeck Sent: Friday, August 08, 2003 5:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Killing runaway PBX How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill PID. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
Will there be a way to delete messages from email. I love getting voicemail in wav to my email, but I hate having to delete them when I call in to get my messages. If we could add a link and have a cgi delete the messages that would be a nice time saver. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman Sent: Wednesday, July 30, 2003 4:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail2 Wish List On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote: Ah, now that you mention it, I implemented this in my patch also and then forgot about it: messages that are too short (less than 3 seconds) or all silence Perhaps this should be configurable? Yeah, I suppose it should. I added minlength (in seconds) and removesilent (yes/no) as [general] options, with 3 and yes as defaults, respectively. Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
I'm not running a high security solution so an link in w/a MD5 encrypted path name or guid would be sufficient. I don't want to enter a password every time. Can Voicemail2 save the files by unique file name? I'd be happy to write the cgi that deletes the message or marks it as read. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Benjamin Miller Sent: Wednesday, July 30, 2003 12:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail2 Wish List Actually, this is a much bigger task than you would imagine. I have not had time to complete an Unified Messaging component to voicemail, but I would see this as an admiral goal. Most modern voicemail systems have some kind of way to delete or mark the voicemail as read when the message is deleted or read from either telephone or e-mail. The biggest hurdle I have come across for this is how does the user enter their e-mail password into a place where asterisk can use it to log into a users mail box an actually use it as the sole repository for mail messages. I see the tasks that need to be completed are: A) abstract file storage and manipulation in voicemail2 to allow an imap or other type (sql?) of storage plug-in rather than dependency on a specific file system. B) an interface to allow the end user to _securly_ enter the username and password that will be used by asterisk to access the file store. It needs to be secure so that people who have integrated passwords like Exchange/AD aren't passing the keys to the kingdom over plain text. Just my 2 cents worth. Ben -Original Message- From: Todd Lieberman [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail2 Wish List Will there be a way to delete messages from email. I love getting voicemail in wav to my email, but I hate having to delete them when I call in to get my messages. If we could add a link and have a cgi delete the messages that would be a nice time saver. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman Sent: Wednesday, July 30, 2003 4:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail2 Wish List On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote: Ah, now that you mention it, I implemented this in my patch also and then forgot about it: messages that are too short (less than 3 seconds) or all silence Perhaps this should be configurable? Yeah, I suppose it should. I added minlength (in seconds) and removesilent (yes/no) as [general] options, with 3 and yes as defaults, respectively. Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
I did the chown and now I get [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script is writable by world., referer: http://asterisk.weichertrents.com/cgi-bin/vmail.cgi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. you still need to make sure nobody has read/write permission on /var/spool/asterisk/vm/$MBOX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Lieberman Sent: Wednesday, July 30, 2003 3:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems I did the chown and now I get [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script is writable by world., referer: http://asterisk.weichertrents.com/cgi-bin/vmail.cgi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how do I do s extensions with PRI
Put the _X below the first 4 extensions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 5:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how do I do s extensions with PRI I would like to know how to define the s extension when I have an incoming PRI line? Currently I have 5 incoming DID numbers. Four of these DID numbers I have going to specific extensions, the fifth number which is the main number I wish to go to a background sound where callers can hear message, get directory, dial extension, whatever. I see that the way to normally do this would be to define s extensions and then step up the priorities for each action I wished to be taken. However, with the PRI line it seems that I can't use the s extension. I can use exten = _X. but this screws up the other four DID numbers which I have going to specific extensions. Is there a way with a PRI that I can define an s extension or something like it to save from having to type an entire 10 digit string multiple places? Thanks for any suggestions. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog commands
Try setting the flash timing longer, if it's too short * won't realize you hit flash. On my phone I have a switch that configures the flash timing. Check if your phone has one. TL -- Todd Lieberman 800-675-3078 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Tyndall Sent: Tuesday, July 15, 2003 4:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Analog commands Hi, When I use the analog phone connected to Zap/1 how do I transfer hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Transfer Conference
Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me Transfer Denied. When I do a conference call I hit the conf key and then dial the next extension. The new call connects and I hit conf again but the calls do not get bridged. Any Suggestions? I'm using the config files from http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco.com/ Thanks, TL -- Todd Lieberman 800-675-3192
