[Asterisk-Users] Hangup in AGI
If I call the Hangup command from AGI directly of via EXEC Hangup it does not work. If shows on the console but it does not hangup. It continues on to the rest of the priorities in the dialplan.. ___ 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] errors on Pri
I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! J Any ideas. Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 n Z What is event 6? Read on 141, is that a channel in zaptel.conf?? Oh and if an error is unknown why give it a number (500)??? I am running CVS from about a month ago. I updated the CVS that was running before but it did not solve the problem. This machine was running before with a T40O (4 port T1) and a T1000 (single port T-1) I pulled the T1000 and put it in another machine for a month or so to do some testing over IAX between machines. This span does talk to the PSTN via PRI. After I put it back in all the problems started. Could this be a case where the PCI PnP has remembered the card and is causing a problem. All the usual stuff under /proc looks good and the cards are NOT sharing any interrupts. I get no other error messages that I can find. Ztcfg vvv loads without errors and dmesg and all the other logs in /var/log look fine. If someone could tell me what these errors are that would be awesome and I would be willing to create and maintain the What The F is this Error page on the Wiki!
[Asterisk-Users] VideoMail
Since * does video over sip has anyone tried to configure voicemail2 to be able to leave a video message? ___ 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] SIP Friends and MySql
What is the difference b/w USE_MYSQL_FRIENDS=1 and USE_SIP_MYSQL_FRIENDS=1 Am I to think that this replaces the entrys in sip.conf for the registering clients?? If so, I am hosed as I cannot get a ATA-186 to register via MySql, but if I leave the config in sip.conf all is well. Could someone send me one record from their sipfriends table that works??? I see that there is no place to specify nat=yes, host=dynamic, etc. in the table, or am I just barking up the wrong tree. Barking, and looking for a new tree.. Alex
[Asterisk-Users] Somewhat on topic but not * specific..
I have seen many postings today about the choppy sound problem. Some of these problems were fixed with the recent change to rtp.c committed today. However in VoIP we usually do not have control of the quality of the data pipe we travel over. I know there are tools that show sip proxies traversed, how the IP packets reach to the desired endpoint. (traceroute) but is there anything that can be used to 'rate' or 'certify' that a route to a given endpoint has the bandwidth, speed, lack of contention that would make for a good VoIP call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs
I will respond off list, to conserve bandwidth. I feel that this is VERY ON-TOPIC. If this continues, we will be faced with customers shying away from Linux due to the whole FUD factor (fear, Uncertainty, and Doubt). We have all made a commitment be it financial or simply time; we all have an interest in this. Laws can protect but they can also smoother. For a clear example look at the PSTN telecom business in the US. It is a lawyer's dream!! I understand that we all have different views of laws and the way the affect us. However, having something or rather a company like SCO change the rules simply because they want the 'best for the shareholders' scares me. Message: 3 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SCO finds someone to pay!!! Date: Wed, 3 Mar 2004 10:56:04 -0600 Reply-To: [EMAIL PROTECTED] On Tuesday 02 March 2004 22:30, Alex Lopez wrote: I don't believe this!! SCO got some one to pony up 7 figures!! Please don't post off-topic crap like this. I get enough of this on other lists. -Tilghman --__--__-- ___ 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] Sorry about the post, meant to be off-list not on.
Hit reply but did not change address!! None the less, I understand your point and respect it. All I ask is for the same respect. Alex Message: 6 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8 msgs Date: Wed, 3 Mar 2004 12:03:50 -0600 Reply-To: [EMAIL PROTECTED] On Wednesday 03 March 2004 11:47, Alex Lopez wrote: I will respond off list, to conserve bandwidth. Which you didn't do. I feel that this is VERY ON-TOPIC. If this continues, we will be faced with customers shying away from Linux due to the whole FUD factor (fear, Uncertainty, and Doubt). Then post it to a generic Linux list. This list is about using Asterisk specfically, not Linux generally. There are more than enough lists out there for which such a post is on-topic. Please take your non-Asterisk posts there. I don't think there's a single user on this list who cares about the SCO issue who hadn't already heard this news elsewhere. In any case, this list does not need to be the central ground for general Linux news. -Tilghman ___ 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] SCO finds someone to pay!!!
I dont believe this!! SCO got some one to pony up 7 figures!! http://www.eweek.com/article2/0,1759,1541140,00.asp
[Asterisk-Users] I finally did IT!!!! Internal dial tone
After having this bother me for a while and trying to do it
via the source I finally got * to give me a different dial tone for internal calls
versus an outside PSTN call.
I have included the dial plan from extensions.conf.
Background info:
Inside is the context defined in my Zapata.conf file as the
context to use for the zap devices.
Allextensions is just that, all my internal extensions.
Asttrunk is simply our main asterisk server that is
connected to the PSTN. I throw all the outbound calls toward this machine and
have the dial plan setup there. You can simply and your trunk
configurations here if you like. Matching here would be the whole number to be
dialed without the 9.
I had to change the dial tones in indications.conf under [us]
to be something different I am using the au tones as ii is most like the inside
tone that I am used to on my other PBX.
___
;
; Rest of extensions.conf snipped
;
[inside]
include = allextensions
exten = 9,1,Goto(outsidedialtone,s,1)
[outsidedialtone]
exten = s,1,Playtones(350+440) ; US standard
dialtone from indications.conf
exten = _X,1,SetVar(FIRSTNUM=${EXTEN}) ;
Had to get the first digit dialed and hold on to it!!
exten = _X,2,StopPlaytones()
exten = _X,3,Goto(outgoingdial,s,1)
[outgoingdial]
exten = s,1,NoOp() ; needed b/c I send the call here via
Goto
exten = _X.,1,Goto(asttrunk,${FIRSTNUM}${EXTEN},1) ; This
is where the rest of the number gets read sans dial tone.
[asttrunk]
exten = _X.,1,Dial(IAX2/[user]:[EMAIL PROTECTED]/${EXTEN})
___
CAVEOTS
I have not changed any source files in order to do this,
(that speaks wonders for the flexibility of the dial plan!!) However,
this does require changes to the default install. I know that it is a
little bit of a kludge, but it does work!!!
Please feel free to improve, comment, or simply use.
Re: [Asterisk-Users] I finally did IT!!!! Internal dial tone
Good point Brian, but ignorepat only gives you the SAME dial tone not a
different one!!!
I would love to see an option for ignorepat that would do this!!!
Say ignorepat = 9,Playtone(350+440)
Alex
Date: Tue, 10 Feb 2004 16:36:05 -0600 (CST)
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] I finally did IT Internal dial tone
Reply-To: [EMAIL PROTECTED]
If its all zap what was wrong with ignorepat? does it not do this ? I
may
be wrong.
bkw
On Tue, 10 Feb 2004, Alex Lopez wrote:
After having this bother me for a while and trying to do it via the
source I finally got * to give me a different dial tone for internal
calls versus an outside PSTN call.
I have included the dial plan from extensions.conf.
Background info:
Inside is the context defined in my Zapata.conf file as the context to
use for the zap devices.
Allextensions is just that, all my internal extensions.
Asttrunk is simply our main asterisk server that is connected to the
PSTN. I throw all the outbound calls toward this machine and have the
dial plan setup there. You can simply and your trunk configurations
here if you like. Matching here would be the whole number to be dialed
without the 9.
I had to change the dial tones in indications.conf under [us] to be
something different I am using the au tones as ii is most like the
inside tone that I am used to on my other PBX.
___
;
; Rest of extensions.conf snipped
;
[inside]
include = allextensions
exten = 9,1,Goto(outsidedialtone,s,1)
[outsidedialtone]
exten = s,1,Playtones(350+440) ; US standard dialtone from
indications.conf
exten = _X,1,SetVar(FIRSTNUM=${EXTEN}) ; Had to get the first
digit dialed and hold on to it!!
exten = _X,2,StopPlaytones()
exten = _X,3,Goto(outgoingdial,s,1)
[outgoingdial]
exten = s,1,NoOp() ; needed b/c I send the call here via Goto
exten = _X.,1,Goto(asttrunk,${FIRSTNUM}${EXTEN},1) ; This
is where the rest of the number gets read sans dial tone.
[asttrunk]
exten = _X.,1,Dial(IAX2/[user]:[EMAIL PROTECTED]/${EXTEN})
___
CAVEOTS
I have not changed any source files in order to do this, (that speaks
wonders for the flexibility of the dial plan!!) However, this does
require changes to the default install. I know that it is a little
bit
of a kludge, but it does work!!!
Please feel free to improve, comment, or simply use.
--__--__--
___
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[EMAIL PROTECTED]
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[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2719 - 10 msgs
How about a PCMCIA Zapata interface?? Asterisk and its strength kick ass as a test unit. Can't do some of the things a T-byrd can do but the Telco techs freak when you tell them its your PBX!!! ) 10. Re: The Smallest Asterisk Server Ever? (Panny Malialis) Message: 10 From: Panny Malialis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? Date: Tue, 3 Feb 2004 20:58:17 - Reply-To: [EMAIL PROTECTED] I cant wait to see the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlugb2003_12/0094.html I've tried removing the b with no luck Anyone know what the link should be ? Thanks Panny - Original Message - From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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 End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Asterisk-Dev] Asterisk Development Updates
My continued thanks go out to you guys and the rest of the Asterisk community. Asterisk HAS grown quite a bit since I first used it over a year ago, and ALL the credit goes to the community. I look forward to the new tree and will continue to CVS off of the current branch so that I can stay on the cutting edge. Once again thanks for all your time and efforts in making this a great, powerful and unique product. Alex Lopez -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Thursday, January 08, 2004 4:45 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Dev] Asterisk Development Updates Prompted by the recent discussion on the mailing list regarding the Asterisk development and release process (or lack thereof), John Todd, Thorsten Lockert, Brian K. West, and myself have put together a plan to address the most significant two legitimate concerns that have been expressed regarding these processes. Specifically: Concern #1: Asterisk release schedules and path to 1.0.0 Asterisk version 0.7.0 will be released by Monday Jan 12, 2004. Later that week, we will create a stable branch from which eventually 1.0.0 will be tagged. Only bug fixes will go into the release branch, while feature requests and bug fixes will continue to go into the head branch. If you are currently using CVS asterisk on a production server, we suggest that you move to the new stable CVS branch when it becomes available. Instructions for using the new stable CVS branch will be made available on asterisk.org next week. Snapshots of the stable branch will also be made available periodically as Asterisk 0.9.x for those not using CVS. If you wish to remain on the cutting edge, you may leave your system using the head CVS as it is currently. Concern #2: Slow integration of bug fixes and feature requests into CVS With the assistance of John Todd and Brian West, we have added documentation about how the bug tracker operates, available at http://www.digium.com/bugtracker.html. This document should help new users understand how the process of submitting bugs works, how to properly follow up on bugs to be sure they get applied, and how to contribute to the bug tracking process as a Bug Marshal, thus accellerating the process. In addition, I am commiting 5-10 dedicated hours of my own time per week to work with Bug Marshals on reviewing bugs, patches and feature requests. Conclusion: Hopefully these steps will help improve the quality and stability of the Asterisk code, and make it easier for people who wish to contribute to Asterisk to do so, while maintaining Asterisk's availibility to continue to advance new features and applications. Mark, John, Thorsten, and Brian ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Different Dial tones for internal and external.
On systems even key systems it is customary to have an internal
dial tone.
Since Asterisk simply ignores the 9 and keeps the tone going
it is hard to tell for some new users if they can make a call.
My first idea was to change the generated dial tone via
source. Then if the user presses 9 go to a different context where I would record
about 30 seconds of the normal dial tone and then let them enter the numbers to
dial. Something it this:
[internal]
exten = 123,1,macro-stdexten(blah,blah,blah)
exten = 124,1,macro-stdexten(blah,blah,blah)
exten = 125,1,macro-stdexten(blah,blah,blah)
exten = 9,1,Goto(trunkgroup,s,1)
[trunkgroup]
exten = s,1,DigitTimeout,5
exten = s,2,ResponseTimeout,10
exten = s,3,Playback(bell-dialtone)
exten = _X.,1,Dial(Zap/g2/${EXTEN})
It Works but there HAS to be a better way!!!
Maybe instead of ignorepat a changetonepat in the context.
How do others do this or am I the first
[Asterisk-Users] Problem with SIP and DOS attacks...
There was a tread that I googled for and could not find about Asterisk being open to SIP DOS Attacks. I have a customer whose machine was hammered last light by traffic on its SIP port causing the OS to use up its resources. Namely number of open files. The discussion was around the fact that the Sip protocol answers requests without regard to authentication. Can anyone comment on this
[Asterisk-Users] Pager support
I have a request from a customer to provide several levels of paging when someone leaves a VM. On other PBX's VM systems it is common for the system to 'dial out' to a pager wait a few seconds and then send a DTMF sequence that is usually the mailbox number, or sometimes it will call a remote phone number and ask for a password and let them listen to their VM message. I have set something up using cron and shell scripts with app_queuecall but I was looking for a more elegant solution. Does anyone have something like this already coded. I will gladly post my scripts if anyone asks. I have seem some discussion of an application called app_hasvoicemail has anyone seem this?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Following completion when Dialing.
I saw that if I add a c to my Dial string as follows: exten = s,3,Dial(Zap/g2c/18005551212) That it will not consider that call as answered until the called party presses #. When the number dialed picks up does a bridge of the call immediately instead of waiting for the # key. I am using a PRI, does that make a difference
[Asterisk-Users] INFO: How the T410P sets the number of channels per span
After speaking with Martin @ Digium, I have the following answers. The driver Wct4xxp determines the number of channels by the signaling type set in the /etc/zaptel.conf file. For example if all the spans used b8zs,esf your spans would look like this: Span 1 Zap/1 - Zap/24 Span 2 Zap/25 - Zap/48 Span 3 Zap/49 - Zap/72 Span 4 Zap/73 - Zap/96 However if you have an E1 intermixed say on span 2 you channels would be: Span 1 Zap/1 - Zap/24 Span 2 Zap/25 - Zap/54 This is because the signaling s\is set to E1 Span 3 Zap/55 - Zap/78 Span 4 Zap/79 - Zap/102 So it is the driver that sets up the devices in the /dev/Zap directory, This could be a gotcha for newbies as the channels are all contiguous and the span separations are on numbers other than the standard 24 or 30. Of Course this is just an example and in a real life situation you would probable start with an E1 or end with an E1 instead of putting it smack dab in the middle. I hope the this helps. Message: 2 Date: Thu, 24 Jul 2003 01:07:54 -0400 From: Alex Lopez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] T410P and zaptel.conf Reply-To: [EMAIL PROTECTED] One a t400p I know that I have 24 channels per port for a total of 96. = However the T410 card allows for E1 as well as T1 lines. How does it = determine how many channels per port. For a more specific question. Would the first Zap device on the second = port be Zap/25 or Zap/30 when using a T1?? I looked for docs on this but found nothing.. =20 Other questions: Is the electrical interface the same for a E1 as a T1?? How does the card know which is which? Is it by the span def in the = /etc/zaptel.conf file? Has anyone seen a technical document on this card??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T410P and zaptel.conf
One a t400p I know that I have 24 channels per port for a total of 96. However the T410 card allows for E1 as well as T1 lines. How does it determine how many channels per port. For a more specific question. Would the first Zap device on the second port be Zap/25 or Zap/30 when using a T1?? I looked for docs on this but found nothing.. Other questions: Is the electrical interface the same for a E1 as a T1?? How does the card know which is which? Is it by the span def in the /etc/zaptel.conf file? Has anyone seen a technical document on this card??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs
I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E911 and asterisk Reply-To: [EMAIL PROTECTED] Isn't that enough to set up a proper Caller ID NAME ? Martin On Mon, 21 Jul 2003, Alex Lopez wrote: I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did they handle 911 services??? For now I am quoting a one port FXO card to be placed in each location, that will in turn connect to a POTS line. However, even though we can use it for the alarm system and it is a kind of insurance I would like to do away with it! ___ 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] Re: SIP Authentication bug?
I have seen this, along with other strange SIP auth issues. I just thought that you HAD to stop and restart * for the changes in the sip.conf file to be reread. I also have not been able to get auth to work. If I put a password in the Windopws Messenger field asterisk does not authenticate. I have tried the plaintect and insecure options in the config file. But, Nothing... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, July 21, 2003 4:44 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #873 - 16 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] Date: Mon, 21 Jul 2003 12:32:23 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Authentication bug? Reply-To: [EMAIL PROTECTED] Might enter it in the new Asterisk bug tracker Mark On Mon, 21 Jul 2003, Tan Aks wrote: Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs
I asked on the IRC channel last night and was told the IAXTEL had been down for a few months now. It had a very poor uptime.. Maybe someone can tell us why the uptime was so poor. Alex Message: 9 Date: Wed, 9 Jul 2003 01:05:00 +0200 From: Paul Cheng [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAXTEL toll-free Reply-To: [EMAIL PROTECTED] Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. www.iaxtel.com also appears dead. Is this the server problem again or is it my config? Haven't been able to find any references in the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIp Registration
I use Windows Messenger ( I duck as to let the hurled penguins barely miss my head J ) and I am able to place and receive calls. So what is the problem you ask??? If I specify a password in the password field of WM I get a Proxy Authentication Error during SIP debug and I am not able to connect until I remove the secret=blah and do not specify a password. Has anyone had this problem before???
[Asterisk-Users] Loaded latest CVS and get Broken PIPE!!!
I updated to the latest CVS about 4 hours ago. If I let asterisk run, connect a few times via r after about one hour the system does not let me in and it starts sucking up resources. One time it spawned about 100 processes. Had to do a reboot. This time it just did not let me in and started using up CPU time. Any ideas
[Asterisk-Users] Call Back
We have Nextel Cell phones where incoming calls are free!
I would like to call a DID number on my Asterisk server,
have it grab my caller ID, not pick up, wait a few seconds, and call me back.
I have already set up qcall to do this and pass it to a
context that asks for a password via authenticate, and is limited to dialing in
my local area so I am not worried about fraud.
I am at the point where it all works except I do not know
the variables in extension.conf {$CALLERID} is the whole strings
including name!! I want just the number.
I could also use this to set up a ANI announcement where you
call the * box and it would use SayDigits to read the number you are calling
from. I searched the archives via google and found nothing.
Anybody got any ideas???
[Asterisk-Users] HELP, I am a newbie.
OK this is what I have done so far. I compiled and install all packages. No errors (wow!) I have configured /etc/zaptel.conf with one span on a t1000 card. as follows span=1,0,0,esf,b8zs I have loaded the demo configs, I have no alarms on my Atlas 550, however when ever I place a call to the unit I get a unavailable from the ISDN side. What am I doing wrong??? All I want at this point is to have ALL the channels on the PRI be two-way, meaning that I want to call into * and have it answer. Later if I want to place a call, grab the next available B-chan and place call. I will not have a channel bank connected to this for analog phones, I will be a VOIP and IVR only box for now. I few pointers in the right direction would be GREATLY appreciated. Once I understand the thinking behind the configs I can get going. THANK YOU!!! alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FAQ
Is there a FAQ for this group??? I have many questions. Most of them I am sure have been answered time and time again. examples: How do I configure an ATA 186 to work with * I have an t1000 hooked up to an Adtran Atlas, but no work. How can I write ACD and IVRs? Is there a error log for asterisk, I try to start it but it just exits. No errors, no prompt, nothing.. etc. etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
