Re: [Asterisk-Users] reverse the selection order of zap channels for outgoing calls
On Thu, 16 Sep 2004, Christopher L. Wade wrote: The subject says it all. Is it possible, code wise, configuration wise, at all - to reverse the order in which the available zap channels are used for *outgoing* calls? Code wise, I looked at the channel structure and it appears as though there is only a next pointer, not a previous pointer, so to 'easily' to this in the code would require a change to the code that reads in zapata.conf? I know I could simply plug the wires in 'backwards' for POTS lines, but I was just wondering if this could be done otherwise. Definitely not a asterisk-dev post, keep this stuff on asterisk-users please. With Zap groups you have a few ways to handle it. Zap/g1 is group 1 starting from the beginning each time Zap/G1 is group 1 starting from the end each time Zap/r1 is group 1 round robin from the beginning Zap/R1 is group 1 round robin from the end James ___ 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] Can not make progdocs
The 'dot' program comes with the GraphViz package. You can also edit the contrib/asterisk-ng-doxygen file and set HAVE_DOT = NO (the default is YES) James On Sun, 25 Jul 2004, Lyle Giese wrote: Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later doesn't seem to work. I have been through several installs getting things to work. This seems to be the last 'bug' up to now. ___ 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] Fax Detection
On Thu, 8 Jul 2004, Ryan Courtnage wrote: AFAIK it's the digium card that _detects_ the fax, and allows the call to jump to the 'fax' extension. So fax _detection_ is a function of the card/driver .. and using the 'fax' extension requires the use of a digium card. SpanDSP just talks fax .. I don't think it actually does any detection. It's not the card that detects the fax. Its the builtin code in asterisk that does it (dsp.c). chan_zap.c is currently the only channel driver that uses the faxdetection but in theory it could be enabled/used in other channel drivers as well James ___ 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-Exec Problem
On Mon, 28 Jun 2004, Tom Daly wrote:
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
The proper usage would be:
$AGI-exec('Record', $vmfile:wav|30);
I guess it isn't clearly documented in my code/examples so I'll try to add
some in before the next release. When it was implemented the | was the
only seperator in asterisk, it wasn't until many months later that the
(,,,) args were implemented
James
http://asterisk.gnuinter.net
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Re: [Asterisk-Users] Security Vulnerability in Asterisk
This was fixed in cvs HEAD and stable on 4/13/2004 and a new source release was made at the time (version 0.9.0) I'm not sure why it would be brought up on a recent newsletter, it was discussed in here (or maybe on -dev) sometime around 4/15/2004 James On Mon, 28 Jun 2004, Jim Rosenberg wrote: The following is pasted from SecurityFocus Newsletter #254: - Asterisk PBX Multiple Logging Format String Vulnerabilities BugTraq ID: 10569 Remote: Yes Date Published: Jun 18 2004 Relevant URL: http://www.securityfocus.com/bid/10569 Summary: It is reported that Asterisk is susceptible to format string vulnerabilities in its logging functions. An attacker may use these vulnerabilities to corrupt memory, and read or write arbitrary memory. Remote code execution is likely possible. Due to the nature of these vulnerabilities, there may exist many different avenues of attack. Anything that can potentially call the logging functions with user-supplied data is vulnerable. Versions 0.7.0 through to 0.7.2 are reported vulnerable. - What is the status of CVS-current with respect to this? I don't remember seeing any discussion of this issue here; apologies if I missed it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Security Vulnerability in Asterisk
On Mon, 28 Jun 2004, Jim Rosenberg wrote: I have to say -- with somewhat less vehemence -- that I'm another user who sure never noticed that the stable release of Asterisk had moved from 0.7.2 to 0.9x. This should have been an important announcement on *SEVERAL* security grounds. As of 0.7.2, the recommend version of channel H323 had some very serious vulnerabilities that the OpenH323 folks had fixed months previously. It's nice to know in the case of these format string problems that they were in some sense addressed promptly, but we're not all subscribed to the dev list. A vulnerability that is fixed in CVS head but not back-patched to stable *is not fixed* as far as a large percentage of the user base is concerned. It was fixed in CVS head and stable and at the same time 0.9.0 was released. The existance was noted in the ChangeLog as well that comes with asterisk Asterisk 0.9.0 -- Logging fixes (fixes remote DoS) -- Fixes from the bug tracker -- ADPCM Standardization -- Branch to Stable CVS I'm not sure if there was an announcement posted to the lists about the code release, but it was definitely updated on the asterisk.org page and the wiki James ___ 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] CDR for transfered calls
On Tue, 15 Jun 2004, John Todd wrote: Not all Asterisk channel types lose control when they step out of the media stream. SIP devices will notify the proxy server (Asterisk, in our case) that a call has terminated, so even though the media stream never went through the proxy (Asterisk). This is an important distinction between IAX2 and SIP - media and control messages are not tightly linked with SIP as they are with IAX2. Until IAX2 has some type of backwards path notification, there is no method (to my knowledge) that IAX2 can notify the origin servers that the call has been terminated. The only way for any server in the path to know the status of the call (with IAX2) is to not transfer the call away from itself, thus bearing the full load of the media stream and the control channel. I would be happy to learn of other's experiences in managing this issue, as it is quite important (mandatory, really) in any type of managed service environment. If you depend on behavior of an unspecified client your asking for trouble. It would be pretty trivial to get a client to send back this 'Call Completed' message right after a re-invite/native bridge has completed. If you have access to the source you can modify the client to do it, or if you have access to the network you have something proxy'ng the packets and at certain times including or excluding certain messages James ___ 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] catch when no voicemail configured
On Fri, 11 Jun 2004, Michael George wrote: I would like to be able to send calls to voicemail() but then catch the case where this is no VM configured on the extension and then go back to another menu. However, VoiceMail() will return a -1 and then hangup. I would like to trap it somehow. Is that possible? You can call the MailboxExists application before hand to check to see if the box exists. -= Info about application 'MailboxExists' =- [Synopsis]: Check if vmbox exists [Description]: MailboxExists([EMAIL PROTECTED]): Conditionally branches to priority n+101 if the specified voice mailbox exists. James ___ 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] Splicing audio clips into one stream
cat 1.gsm 2.gsm 3.gsm new.gsm works fine James On Wed, 2 Jun 2004, Michael Welter wrote: Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.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] Some (lack of) answers regarding the wakeup call application...
On Tue, 1 Jun 2004, Rob Fugina wrote: On Tue, Jun 01, 2004 at 06:54:02PM +0300, Apollon Koutlides wrote: Rob Fugina wrote: It has occurred to me that the two AGI scripts could be rewritten as real compiled asterisk applications, but then it always hits me that without some kind of cron-line built-in scheduler, or changes to the outgoing call queueing that would allow a call to be scheduled for the future, there would still be that external cron-driven shell script. Ugly. Actually, there's no need for anything like that. Set the file's modification time to the value you require, and watch asterisk do all the dirty work for you. Sounds like a race condition to me... Creating the file, then modifying the timestamp to a future time, hoping that * doesn't grab it away in the mean time... I'll look further into this one, though... The old app_qcall would flock the file, assuming the app that drops the file in would also flock the file then there wouldn't be any problem. It would be trivial to add this same functionality to pbx_spool On occasion I've seen pbx_spool start processing a file before the write has been complete. To get around this I build the file in an alternate directory (on the same filesystem) and then move the file over. James ___ 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] Queue Hold Time
On Thu, 27 May 2004, John Congdon wrote: I know that you can announce the current hold time to the caller, and that this hold time is based on the box car filter... Are there any current plans to include a management type of feature to export actual average, min, and max hold time. Is there already something implemented that I am unaware of? I'm not sure if it contains the information you seek, but a few months back a queue log was added. You can find some docs in asterisk/doc/queuelog.txt James ___ 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] Questions about alarm reporting in Asterisk
On Wed, 21 Apr 2004, Clif Jones wrote: I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most common problems are on the infrastructure network but are reported as phone system problems because that is the piece that the customer directly interacts with. I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. The manager interface sends events when a channel/span goes into alarm. A simple app collecting this data should be able to handle this for you James ___ 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] Intel 536ep as a FXO?
On Tue, 20 Apr 2004, Tom wrote: SBC cancels milliwatt tone generators. -- I called our local SBC CO and asked for a milliwatt tone generator number. He said that SBC decided they were not needed and put out an order to remove them in February. The tech said they have been removed from all SBC COs. :( We are in northern Illinois. This isn't exactly true. SBC might have put the word out to cancel all the numbers, but its up to the end offices to actually do the work. I just tried abot a dozen test numbers and all but 1 worked still. James ___ 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] Milliwatt Quiet terminations
On Tue, 20 Apr 2004, tmpm wrote: If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is quiet term. I put them in that Ericsson AXE-10 in 1984 and they're still there. Oh one more thing nobody has pointed out yet. * comes with an app that can do ths as well. -= Info about application 'Milliwatt' =- [Synopsis]: Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description]: Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) James ___ 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] Upgraded to latest CVS, now no IAX1?
On Tue, 13 Apr 2004, Eric Wieling wrote: Tor Houghton wrote: Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Are you using cvs latest or cvs stable? I thought IAX1 was still in cvs stable, but I could be wrong. To enable IAX1, the following line in channels/Makefile needs to be uncommented. # If you really want IAX1 uncomment the following, but it is # unmaintained # #CHANNEL_LIBS+=chan_iax.so James ___ 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 call manager
On Wed, 7 Apr 2004, Jain, Sonal wrote: I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 Connected to localhost Escape character is '^]'. Asterisk Call Manager/1.0 Then I type Action:Login (enter) Username:sam Secret:sam Then I enter twice I get Response: error Message: missing action in request I am not sure what it means. Thanks You need a space after each header. Action: Login Username: sam Secret: sam from doc/manager.txt: Command Syntax -- Management communication consists of tags of the form header: value, terminated with an empty newline (\r\n) in the style of SMTP, HTTP, and other headers. James ___ 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] Spring VON Wrap Up
On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. James ___ 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] SetCDRUserField actually works?
On Tue, 6 Apr 2004, Leo Ann Boon wrote: I've looked into cdr_csv.c, cdr_pgsql.c, cdr_addon_mysql.c from 0.7.2. So far, only cdr_csv has it as a #define. The others all support userfield by default. One other observation, cdr-uniqueid is the only field that's controlled via #define in all the 3 cdr modules. By default uniqueid is not logged, which I find is counter-intuitive. Any cdr guru out there care to enlighten everyone? uniqueid was added long after the original cdr code was released, so it was added as a define to explicitly enable it. At some point it might end up enabled by default, but being backwards compatible by default is a good thing. James ___ 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] Problem with Manager Originate
On Sun, 4 Apr 2004, Serge Mankovski wrote: Hi I am trying Manager interface for originate a call. This is what I get --- Action: Originate Exten: 555 CallerID: test 6656 Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed What do I do wrong? Check the errors/messages on your console. I suspect you will see some messages about 'unable to create channel SIP/8782'. The Originate failed message pretty much only comes up when there is a problem creating the Channel. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: What failed here?
On Tue, 30 Mar 2004, John Chambers wrote: Another worrying thing that I've noticed: The stuff at the start of the make (that scrolls off the top too fast to read ;-) first does a mkdep, and then these messages appear: cli.c:31:19: build.h: No such file or directory dlfcn.c:40:25: mach-o/dyld.h: No such file or directory dlfcn.c:41:26: mach-o/nlist.h: No such file or directory dlfcn.c:42:28: mach-o/getsect.h: No such file or directory Sure enough, those files don't exist. Some time later, build.h does appear, when the Makefile runs make_build.h. It seems a bit odd that the Makefile would attempt to use build.h before creating it. This looks like a sign of something wrong, but I can't tell what. Any idea how to fix this? Or is it actually a problem? The mkdep simply builds .depend files in each directory of the source tree. make uses this to determine what needs to be rebuilt if one of the header files has changed. There is nothing to worry about at all with that part. It looks like either your CC line in the Makefile has been changed, or perhaps your overriding it somehow. Make sure your using unmodified code. /var/spool/asterisk is a directory not an executable, and make should be calling gcc instead of /var/spool/asterisk James ___ 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] T100P T1 problem (Avaya - asterisk IVR) (update)
On Mon, 29 Mar 2004, Eric Wieling wrote: Jeb Campbell wrote: Anyway, the only stuff off list was trying to debug the connection. 1. With a crossover there is no sync (YELLOW and RED alarms) 2. With standard cable I get a pri error that they think they are the NET, but we are the NET. (This is asterisk 1.0 stable and the directions from voip-info) If they think they are NET then make Asterisk CPE, if they think they are CPE then make Asterisk NET. Another possibility is that the interface is looped back so the NET packets it sees are the ones asterisk sends out and gets looped back James ___ 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] SoftFAX/spandsp
On Sun, 28 Mar 2004, Florian Overkamp wrote: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from /usr/local/include/spandsp.h:40, from app_rxfax.c:29: /usr/local/include/spandsp/arctan2.h: In function `arctan2': /usr/local/include/spandsp/arctan2.h:44: warning: type mismatch in implicit declaration for built-in function `fabs' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:185: too few arguments to function `ast_set_read_format' app_rxfax.c:195: too few arguments to function `ast_set_write_format' app_rxfax.c:199: too few arguments to function `ast_set_read_format' app_rxfax.c:247: too few arguments to function `ast_set_read_format' app_rxfax.c:253: too few arguments to function `ast_set_write_format' make[1]: *** [app_rxfax.o] Error 1 The ast_set_read_format and ast_set_write_format functions have been changed in CVS head (but not stable), to include a flag if the channel should be locked. James ___ 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 as an IETF Standard?
On Wed, 24 Mar 2004, Olle E. Johansson wrote: An informational RFC documenting the protocol would be a good start, it would make it more open but not an IETF product. Security specialists would get something to read and analyze. A VOIP protocol with RSA authentication, implemented today. Is there any IAX2 document that could be a basis document somewhere? There is the beginnings of a whitepaper at: http://www.cornfed.com/iax.pdf I think putting together a RFC for IAX2 is a great idea, but who has the time. I'd be willing to help in any way if someone was planning on it though. James ___ 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] Should List be Moderated?
This was posted last year by Mark. I figured I'd repost it to refresh peoples memories. Please stop posting commercial postings and announcements to the *-users and *-dev. Let's self moderate so the list doesn't have to be moderated James -- Forwarded message -- Date: Mon, 7 Apr 2003 18:43:17 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Should List be Moderated? In light of recent flame baits and advertisements sent to the list, I would like to seek opinions of list members on making the list moderated. I certainly don't have time to moderate the list myself, so I would suggest giving at least a half dozen, maybe more, people the ability to approve posts to keep it flowing quickly. Moderators would be asked just to approve/disapprove based upon a specific list of characteristics. Among characteristics that *could* be considered: * Posts should not advertise products, especially not those unusuable under Asterisk * Posts should not contain profanity * Posts should not simply be me-too's * Arguably, maybe something related to flame baits Any comments on any of these rules, or suggestions for others, that would make the list more valuable? Mark ___ 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should List be Moderated?
And thus Asterisk-Biz was born. (http://lists.digium.com/mailman/listinfo/asterisk-biz) On Thu, 18 Mar 2004, Panny Malialis wrote: So give us a commercial list. Please :) Panny - Original Message - From: James Golovich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 5:10 PM Subject: [Asterisk-Users] Should List be Moderated? This was posted last year by Mark. I figured I'd repost it to refresh peoples memories. Please stop posting commercial postings and announcements to the *-users and *-dev. Let's self moderate so the list doesn't have to be moderated James -- Forwarded message -- Date: Mon, 7 Apr 2003 18:43:17 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Should List be Moderated? In light of recent flame baits and advertisements sent to the list, I would like to seek opinions of list members on making the list moderated. I certainly don't have time to moderate the list myself, so I would suggest giving at least a half dozen, maybe more, people the ability to approve posts to keep it flowing quickly. Moderators would be asked just to approve/disapprove based upon a specific list of characteristics. Among characteristics that *could* be considered: * Posts should not advertise products, especially not those unusuable under Asterisk * Posts should not contain profanity * Posts should not simply be me-too's * Arguably, maybe something related to flame baits Any comments on any of these rules, or suggestions for others, that would make the list more valuable? Mark ___ 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 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] Cisco 7960 and short delay before voice starts after ring.
On Thu, 11 Mar 2004, Iain Stevenson wrote: I hacked the Wait command to wait in increments of 100ms. The 7960 needs about 300ms delay after answer to play the sound properly. ATA186's work fine without any delay for me. A finer grained 'Wait' would be helpful in developing workarounds for this sort of problem. As of 3/4/2004 in cvs head and stable the Wait application has accepted time with fractions of a second. So 0.1 would be 100ms, 0.3 would be 300ms, etc. James ___ 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] g726 Codec w/ GrandStream 1.0.4.50 Firmware
On Sun, 7 Mar 2004, Greg Boehnlein wrote: On Mon, 8 Mar 2004, Master Abi wrote: Upgrade to the latest CVS and ast_rtp_read/write warnings will disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works great. Hmm.. when was this fixed? I'm running a CVS version that was pulled and built this morning, however I believe that I'm running the 1.0_stable branch on this box. The G726 codec is not in the 1.0 stable branch, only in the HEAD branch of CVS James ___ 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] g726 Codec w/ GrandStream 1.0.4.50 Firmware
On Sun, 7 Mar 2004, Greg Boehnlein wrote: On Sun, 7 Mar 2004, James Golovich wrote: On Sun, 7 Mar 2004, Greg Boehnlein wrote: On Mon, 8 Mar 2004, Master Abi wrote: Upgrade to the latest CVS and ast_rtp_read/write warnings will disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works great. Hmm.. when was this fixed? I'm running a CVS version that was pulled and built this morning, however I believe that I'm running the 1.0_stable branch on this box. The G726 codec is not in the 1.0 stable branch, only in the HEAD branch of CVS Yes, unless you apply the patch from bugs.digium.com! ;) Which I did, about 10 minutes after it was posted. ;) The only patch that was posted was for the file format G726, not the codec g726. James ___ 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] new2agi -php
On Sat, 6 Mar 2004, Doug Harris wrote: * telles me Error in Argument 1, char 3, option not found. . script can be run from command line. Appreciate some help to get going here. I can't find any error like this in the asterisk source code, perhaps your error is coming from php? James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Outgoing parallelism
Its the mtime of the file James On Sun, 29 Feb 2004, Eric Wieling wrote: You can manage when the call starts by setting the atime (or is it ctime?) of the blah.call file. See my sample script at www.fnords.org/~eric/asterisk It's the Callback script. You can also manage it WITHIN the AGI application run by the blah.call script, of course. On Sun, 2004-02-29 at 10:49, Bill Michaelson wrote: Thanks, Scott. I'm in a general exploration mode, but I do have a small broadcast application in mind. My limited experimentation leads me to suspect that there is no queue management at all. I was testing with only a single call file just minutes ago, and the system tried to redial the destination as a retry (60 second interval had been spec'ed), even though the first call was still in progress! I suppose I will have to manage throttling with some kind of completely external process, which is likely to be cumbersome. For the immediate application, and given my current facilities, single threading will be adequate (and necessary), but from what I've seen, even this could be challenging. If I put together anything generally useful, I'll share it. From: Scott Stingel [EMAIL PROTECTED] Hi Bill- I've built some load testers for asterisk, using the outgoing call facility. It's been a little while, so you may want to test this yourself, but I recall finding a couple of problems: (a) I don't think it manages queuing very well if there are a limited amount of outbound resources. For example (again, from memory), if you define a group (g9 for example) of two lines for use in outbound calling, it works fine if the number of outbound calls to be made at any moment never exceeds 2. A third call file in this example, will be grabbed by asterisk, but will fail immediately. So I had to create a mechanism in my Perl script to ensure that the outbound calls actually completed - no easy feat since I couldn't tell when that occurs from the perl script too easily. (b) There was a problem dumping more than about 12-15 outbound calls at once in the outgoing directory, even if there were plenty of channels available to make the calls. Asterisk would grab them but would not process some of them. This is a load-testing scenario, and not too common I realise, but something to be aware of. It didn't seem to matter if I switched to a more powerful processor. These problems occurred using a December release of asterisk - maybe they are fixed now?? Please let me know if you are doing any load testing, and I'll send you some simple scripts if you like. The outgoing facility works fine at lower call volumes, if you stagger the creation of the files in the outgoing directory. ___ 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 -- Eric Wieling [EMAIL PROTECTED] BTEL Consulting ___ 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] A missing argument
On Mon, 23 Feb 2004, Dave Cotton wrote: Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2 with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3 the most 10.0 fails at this point chan_zap.c: In function `handle_init_r2_event': chan_zap.c:4773: error: too few arguments to function `zt_new' make[1]: *** [chan_zap.o] Error 1 line 4773 has chan = zt_new(i, AST_STATE_RING, 0, SUB_REAL, 0); but greping shows that the declaration and other instances have 6 arguments. ML 9.2 is using gcc-3.3.1 whilst 10.0 is using gcc-3.3.2 What worries me is how many other programs in the world have the same type of error and the compiler has missed it? Dave, You if have libr2 installed. I don't believe much work has been done on R2 in quite some time, so it might not be up to date. The channels/Makefile looks for the existance of /usr/lib/libmfcr2.so.1 to set ZAPATA_R2 which is causing those sections of code to be compiled in James ___ 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 Faxing
On Sun, 22 Feb 2004, Andrew Kohlsmith wrote: First, please no HTML email. Not only does it more than double the size of your messages, it is plain bad etiquette for mailing lists. Second, do not reply to a post, erase everything and then type your message -- it breaks threading in a horrible way and buries your question in the middle of another thread, where fewer people can see it. Just click on the [EMAIL PROTECTED] and start a new message. It's faster, easier and works far better. What is the best or simplest method to connect 4 fax machines into a * system? I am going to be testing the fax - TDM400P solution shortly to see how it works. Please keep in mind that if you are connecting a fax machine and you expect it to work, you *MUST* use the ulaw or alaw codecs (everything else is geared for voice and will horribly break faxes or even DTMF tones) and more importantly, you must disable the agressive echo cancellation in the zaptel driver, as it tends to kill the faxes own echo cancellation. Whether you can simply say echocancel=no for the appropriate Zap channels I'm not sure. I don't explicitly disable echocancellation on the channels I use for fax, and zaptel always seems to detect the tone to disable echo cancellation from the fax. I send/receive all my faxes over IAX2 with g711ulaw James ___ 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 Faxing
On Sun, 22 Feb 2004, Barry Fawthrop wrote: - Original Message - From: James Golovich [EMAIL PROTECTED] James Golovich wrote: I don't explicitly disable echocancellation on the channels I use for fax, and zaptel always seems to detect the tone to disable echo cancellation from the fax. I send/receive all my faxes over IAX2 with g711ulaw James How do you connect your fax machine, I'm interested to know? Have you assigned a dedicated channel and local PSTN number to the fax machine, or what did you use? I have my fax machine plugged into an FXS port on my channel bank, plugged into a T400P. I have assigned myself a DID for fax only that I route to myself over IAX2 James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: multicasting conference calls
On Mon, 23 Feb 2004, dkwok wrote: Use the outgoing call feature of asterisk to have the servers join each others conferences. It's very simple. Sorry, I am not quite sure what is the outgoing call feature. Would you please elaborate a bit. Asterisk has a few different ways to generate outgoing calls. I'd suggest taking a look at: http://www.voip-info.org/wiki-Asterisk+auto-dial+out James ___ 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 did not go through
On Sat, 21 Feb 2004, Jim Sneeringer wrote:
[default]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
where
TRUNK = Zap/1Zap/2
which are Digium FXO cards.
It works with
exten =9,1,Dial(${TRUNK}/${EXTEN})
Furthermore, it was working before. To my knowledge, the only thing I
changed to make it fail was to shut down the working test system, move it to
the actual environment, and make it live. I had been testing with only one
of the two CO lines. Maybe I changed something in extension.conf, but if so
I don't know what it was.
The message you are getting is not from asterisk, its from you telco. To
simulate what * is doing in this case, plug a phone into your POTS line
and just pick it up without dialing any digits.
You should be using a zaptel group for this, as Dial'ng the way you
are now won't work properly.
Assuming you are dialing 95551234 this is what would be dialed
Dial(Zap/1Zap/2/5551234)
So it will only actually dial a number when the first Zap channel isn't in
use.
Make sure you have a group set in /etc/asterisk/zapata.conf before both of
your channel definitions. Like this:
group = 2
channel = 1
channel = 2
Then change your TRUNK to:
TRUNK=Zap/g2
James
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Re: [Asterisk-Users] Dialing 800 numbers with VOIP
It appears that iaxtel doesn't support IAX1 anymore, so gnophone should no longer work with it. I don't use iaxtel or gnophone, so someone correct me if I'm wrong. I don't believe I am though. James On Fri, 20 Feb 2004, Mukul Prasad wrote: HI I am trying to run gnophone with asterisk PBX . The problem is that the originating gnophone always shows the status of call Attempting connection while the terminating gnophone goes in the active state and shows 1 call active. The Media also doesn't get through I am also attaching the logs of gnophone. Hope somebody can help me . Thanx 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
RE: [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF
Also since you are using Asterisk::AGI you can register a callback that
gets called when most of the AGI commands return error/hangup.
James
On Thu, 19 Feb 2004, Tim Petlock wrote:
Thanks - that gave me the basis for a couple of google searches.
Near the top of the script I put in
$SIG{HUP} = \exitGracefully;
and I added a subroutine that looks like this:
sub exitGracefully {
exit(0);
}
It now kills itself off without needing to be killed and take * with it.
THANKS!!!
-Tim
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Re: [Asterisk-Users] Room Monitor
On Thu, 19 Feb 2004, Jamin W. Collins wrote: Actually the baby monitors tend to be in the 47Mhz band, but yes they still suck. There are newer models in the 900Mhz and 2.4Ghz range. However, reviews of the 900Mhz models are almost unanimous in declaring them to be worse than the 47Mhz models. While my experience with 2.4Ghz phones indicates that they will trash most 802.11b networks. So, I'm trying to move away from the wireless solution. Mainly, due to the interference between the two locations. The three station FM solution looks promising. That is if it can deal with the stations being on different breakers within the same residence. You may also want to look at a better model of intercom. http://www.fisher-price.com/us/babygear/product.asp?id=17605c=bgm Uses 900Mhz. Several reviews of this model indicate severe static problems. To bring this back on topic. Have you considered leaving a phone with the handset off the base, or speakerphone turned on in the room? Set the zap channel to immediate and send it to a special context. Have the s extension send into a meetme that is talker only and then all you have to do is dial into the meetme and monitor the call. James ___ 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 monitor with Daemontools
On Mon, 16 Feb 2004, Jeremy McNamara wrote: EVIL! Asterisk fork's a new process shortly after starting (unless you run with a console) Find safe_asterisk in the contrib directory. If you start asterisk with -f it won't fork James ___ 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] Digium connectivity issue?
On Sun, 15 Feb 2004, John Fraizer wrote: Brian West wrote: But CVS was alive the whole time! ;) bkw Um, no it wasn't. For all practical purposes, *.digium.com was dead. Why? Because even though there is a second cvs.digium.com out there on a different network, the nameservers digium.com are both on the same network - the network that was down. So, there was no way to actually get the addresses for CVS.DIGIUM.COM. This is true, but no longer the case since other nameservers are now setup. Both nameservers on the same /24 = bad. Not to split hairs here, but this statement isn't necessarily true. If it read Both nameservers on the same physical network then it would be true. I've worked on systems that each of the 2 NS glue records were actually /32s located on multiple servers around the country. So even if part of the network was down, multiple servers are always reachable. Now lets return to our regular programming James ___ 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] Readline readline-devel installation on RH9
On Wed, 17 Dec 2003, Ariel Batista wrote: # rpm -q kernel-source redline redline-devel openssl opessl-devel I have done this but my system reports that redline and readline-devel not installed. How do I install these items without re-installing RH 9 all over again? Also why are these needed? Issue #1: It appears you are using redline and not readline Issue #2: readline is no longer needed for asterisk as of sometime around october of 2002 James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nagios/measurement with Asterisk - any plugins?
On Mon, 15 Dec 2003, Florian Overkamp wrote: At the time of writing it served my purpose, but it had some problems with asterisk becoming unresponsive after a large number of manager logins/logouts. I suppose this has been solved by now, but I stopped testing it heavily since I had not had a large amount of requests for it... I'm pretty certain that I fixed that bug a while ago. IIRC it was caused by not creating the new thread in the detached state. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange variable chopping from AGI's
On Mon, 8 Dec 2003, John Todd wrote: AGI's are resulting in unusual behaviors. Can someone please tell me if this is my inappropriate use of AGI's, inappropriate use of Time::HiRes, or a bug with *: I'd say inappropriate use on Time::HiRes. Microseconds increment from 0 up to 999,999 and when it passes that mark the second count is incremented and microseconds is reset to 0. I call this script twice: #!/usr/bin/perl use Time::HiRes qw( gettimeofday ); ($seconds, $microseconds) = gettimeofday; $hirestime = sprintf(%s,$seconds$microseconds); print SET VARIABLE HIRESTIMESTAMP $hirestime\n; There are tons of ways to do this right, but here are two of them. Change your sprintf to: $hirestime = sprintf(%d%06d, $seconds, $microseconds); This will make it so that microseconds will always be 6 characters long or change it to something like: $hirestime = sprintf(%d.%d, $seconds, $microseconds); So there will always be a decimal place between seconds and microseconds. Assuming your later code can deal with it, this is the way I would do it. start time end time duration (endtime-starttime) 1070917681581683 1070917681942384 360701 107091768107 1070917681968283 301676 1070918477712530 1070918478137011 424481 1070917681788671 1070917681998254 209583 1070917681837624 107091768221563 -963825913616061 - error! Makes sense to me, here is how the numbers look when broken in seconds and microseconds 1070917681.837624 1070917682.21563 James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, John Todd wrote: The problem with TFTP is that it is neither authenticated NOR encryptable by nature. I have no issue with the lack of authentication if the files moved can be encrypted. This is a critically important point: sending out cleartext TFTP (or HTTP, for that matter) files across ANY network is ill-advised. I'll add one potential problem to TFTP in the current internet as we know it. With all the recent worms of this summer it has been many vendors recommendations to block tftp. I've seen increasing number of cable/dsl providers following these recomendations. So over the public internet software/config grabbing with tftp could be a potential problem James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi exit problem
On Tue, 7 Oct 2003, Panny Malialis wrote: Not sure if it's possible to keep the script running after Dial but perhaps you could explain what you're attempting to achieve and there may be a workaround. I want to know how long the call lasted :) Your AGI will continue to run, but after the call has hungup you can no longer exectue any AGI commands. Your verbose will fail, but if you print to STDERR you will see that your script is still running. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
On Tue, 7 Oct 2003, john lawler wrote: But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: I've seen similar issues that seem to only happen when /usr is on a seperate filesystem than the root filesystem. I saw this happen on debian. Because /usr/lib/libtonezone.so.1 (or whatever its called) isn't available when ztcfg is run the command does not work. My solution was to have my asterisk startup script execute modprobe to load the module. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phonecore source
On Mon, 6 Oct 2003, James Coberly wrote: Hi, Trying to compile gnophone and am having a bit of a time finding the source for phonecore. Anyone know of somewhere I can pull the source from? I have a copy of the most recent gnophone source located at http://asterisk.gnuinter.net/files/cvsnightly/phonecore-cvsnightly.tar.gz James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPEEX bitrate?
On Tue, 30 Sep 2003, WipeOut wrote: Whats the default SPEEX bitrate set to in Asterisk? The default bitrate for speex (at this time determined by the speex lib because we don't explicitly set it) is 15k I'm still looking for a good way to implement options for codecs so we can modify these settings. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '.' pattern and non-SIP phones
On Thu, 25 Sep 2003, Andrew Kohlsmith wrote:
Using FWD and accessing it via this extension:
exten = _*8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This works *perfectly* with SIP phones. However with a regular phone
plugged into an FXS card (PhoneJack PCI in my case) the '.' traps the first
number dialled after *8 and tries calling that. I've tried setting a digit
timeout but it doesn't seem to help.
Changing that to
exten = _*8X,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
works, but is hardly optimal, since I plan on changing my dialplan to allow
varied-length numbers for other things.
I can't explain it without looking at the code, and I'm short on time so I
won't go there but the way that works best for me is:
exten = _*8X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
James
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Re: [Asterisk-Users] Festival Problems
On Wed, 24 Sep 2003, Borut Senicar wrote: I have exactly the same symptoms with app_festival and I suspect that send_waveform_to_channel routine in app_festival.c doesn't work correctly. Festival works correctly since it sends wave file to asterisk, which saves it in cache. If I strip app_festival header in that file I can play it. The problem lies in playback of this wave to channel. Ant ideas? I didn't see an extensions.conf snippit that goes along with this, but I'm going to guess that the channel hasn't been Answered before the Festival app is being executed James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
On Wed, 24 Sep 2003, Steven Critchfield wrote: As a phone platform, it may be overkill, but I bet it could drive a TDM400P card and be able to handle GSM compression. The question then again is if it is worth the cost for basically a 4 port asterisk based device like the ATA186? I have a mini-itx board (800mhz) and case (I can look up the part number if anyone is interested), but the older TDM400P has the sound problems with the power supply in there. I've been meaning to contact digium to swap the card out and try a new rev in there but I haven't had the chance. For now I have a T100P working in there great. I originally wanted to get the TDM400P working in there because its a great demo system to show people just what it can do. People are very impressed when you can walk in, plug a box in the ethernet (assuming dhcp), plug a phone into the back, and start making calls via IAX. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
On Fri, 19 Sep 2003, WipeOut . wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message-waiting-indicator thru ZAP interfaces- how to?
On Tue, 2 Sep 2003, John Todd wrote: I'm trying to make the MWI indicators on my client's Vodavi Starplus DHS phones work. The actual signalling - in-band DTMF from the ZAP interfaces directly to the PBX system - works fine. I can manually tell asterisk to send #9610 as DTMF and voila, the MWI on extension 10 lights (or goes out.) The question is, how is this integrated with voicemail, i.e. so that the MWI turns on and off appropriately, when new messages arrive and after a user has listened to their messages? I've checked the last two months of mailing-list messages but found no mention of this situation. Any tips or pointers to online docs would be appreciated. Thanks, Sam P.S. Thanks to Jsmith for the fast, simple answer to my last question re: version number in CVS not updating. I'm afraid that the answer to this, without programming some stuff inside of Asterisk, is uuugly. I suspect it will involve using perl or shell scripts to actually peek inside of the /var/spool/asterisk/vm directories and check things manually, out of a cron job or out of the h context with a System call. Then, a call would be created by the script (see sample.call) - just thinking about this method gives me the willies. The clean way would be to put a tiny call into the voicemail app (or would it be in app.c?) that triggers an outbound call with the appropriate parameters (see sample.call) I was working with someone over the weekend that is working on something like this, and it might even be the same type of system because the DTMF trigger looks similar. The easiest way to do this is with an external daemon that connects to the manager interface. You just need to watch for MessageWaiting events and when you see a change of state trigger an Originate action to dial out and enter the required DTMF. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AgentCallbackLogin
The unique id is generated when a channel is requested, so the uniqueid
will be unique for any call or channel. You should never see the same
uniqueid more than once. The uniqueid is available as a channel var
${UNIQUEID}, as an agi environment var (agi_uniqueid), as an entry in
(hopefully) every management command (as uniqueid), and in the CDR if you
have edited the cdr_*.c files to include it (by default its not for
backwards compatibility)
James
On Thu, 7 Aug 2003, Jim Friedeck wrote:
I don't think that's the same unique id. It changes for each record in
the CDR. I believe the management interface unique id is maintained as
specific to each incoming or 'original' call. Any ideas?
Jim Friedeck
---
Jim Friedeck wrote:
Thanks! I'm trying that now.
Jim Friedeck
Armand A. Verstappen wrote:
out. Is there some way to distunguish them in CDR? I also noticed
the management interface maintains a Unique ID for each call and
lets that call be traced throughout its life in the PBX. Can that
data be added to CDR as well to allow for easier call tracking?
It looks like if you define MYSQL_LOGUNIQUEID in the top of
cdr/cdr_mysql.c and recompile, it will start logging the unique id you
want.
wkr,
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Re: [Asterisk-Users] Fax Handled
On Thu, 14 Aug 2003, Eduardo Goncalves wrote:
I'm using G.711alaw.
My extensions.conf:
===
[globals]
TRUNK=Zap/g1
[sip]
exten = s,1,Background(demo-moreinfo)
exten = fax,1,Dial(${TRUNK}/${EXTEN})
exten = _0.,1,Dial(${TRUNK}/${EXTEN})
exten = _9.,1,Dial(${TRUNK}/${EXTEN})
Is this correct?
The last time I looked at the code, fax would only be detected if they
came in on a Zap channel. So if the fax was coming in on a SIP channel
then it would not work.
James
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Re: [Asterisk-Users] h extension seems to wipe variables?
Once the call has hungup the AGI functions stop working, so executing an AGI from an h extension will not do what you expect. but you can still do all kinds of perl stuff there, so you could stick the info in mysql or somewhere else James On Wed, 13 Aug 2003, Alastair Maw wrote: Hi. I'm trying to do some custom call logging, and I want to call an AGI script from a hangup handler to log call durations and things. Although the script executes, it isn't retrieving variables from the AGI interface. Looking closer, I realised the variables are actually getting unset before the h extension is reached. [foo] s,1,SetVar,foo=bar s,2,Play(audio/a-long-prompt) h,1,AGI(log-call-duration.pl) When I do an $agi-get_variable(foo) from the perl, I get the string noresponse returned. This all works fine if I don't call the AGI from the hangup extension, but from a normal one instead. Does anyone have any idea how I might fix or work around this? It's important for us to log call durations (and other things), which obviously needs to be done when the users hangs up. Storing stuff using the cdr isn't really an option. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording
By default the Monitor resource/app uses the channel name as the filename,
but you can override the filename base. A good choice of the filename
base would be the uniqueid for each channel, fortunately the ${UNIQUEID}
channel variable is available.
So from extensions.conf you can do Monitor(wav,${UNIQUEID}) to record
'wav'.
The uniqueid is available in the cdr struct as well, but it isnt used
right now for backwards compatibility. You can edit cdr/cdr_csv.c and
uncomment the: /* #define CSV_LOGUNIQUEID 1 */ line to get it to log the
uniqueid at the end of each entry.
James
On Fri, 11 Jul 2003, Erik Kendall wrote:
Can Asterisk automatically record all calls to unique
files, like voicemail does with the messages?
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RE: Re[4]: [Asterisk-Users] Asterisk PBX Billing
The mysql schema is available in the doc/cdr_mysql.txt file (from the asterisk source dir) James On Thu, 3 Jul 2003, Kim C. Callis wrote: You can find the comma delimited file at /var/log/asterisk/cdr-csv or if you are looking to do some easy querying on a database, you need to create a schema that I am sure someone on the channel has defined somewhere. At that point you clean up the /etc/asterisk/cdr_mysql.conf file to point to the appropriate database and authentication information. Kim C. Callis -Original Message- From: Angelo Sampietro [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 8:07 AM To: Kim C. Callis Cc: [EMAIL PROTECTED] Subject: Re[4]: [Asterisk-Users] Asterisk PBX Billing thanks a lot! can you tell me where can i find more info about the CDR? probably this will be the better way to give to the company a summary with all the phone traffic :) Angelo Thursday, July 3, 2003, 4:37:32 PM, you wrote: KCC There is a CDR (Call Detail Record) which is accessible in two different KCC ways. The first is via a simple comma delimited file which can be parsed KCC and fed into whatever database that you want. The second way is to dump KCC the CDR directly into MySQL, and extract accordingly. So the only trick KCC there is to create a database for billing and create a relationship that KCC will extract from the CDR database. KCC Kim C. Callis KCC -Original Message- KCC From: [EMAIL PROTECTED] KCC [mailto:[EMAIL PROTECTED] On Behalf Of Angelo KCC Sampietro KCC Sent: Wednesday, July 02, 2003 7:06 AM KCC To: Scott Stingel KCC Cc: [EMAIL PROTECTED] KCC Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing KCC i think that the problem could be something more easy: KCC it is possible inside asterisk to log all che calls of all the users KCC and know the timing and the number called for each call? KCC if it is possible to do that, could be possible to make a program KCC that takes this files and generate the costs reading the log KCC informations... KCC so for me the real question is: there is a log of all the phone call KCC that are made by asterisk? KCC Angelo KCC Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. KCC It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a KCC pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it KCC is SS occurring, debiting each account, and the master account, in KCC real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. KCC You SS have to cut them off promptly when the money runs out (big problem). KCC And SS you have to provide call detail and charges to them at the end of KCC each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from KCC the long SS distance carrier, and within the country, major cities often have KCC special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses KCC a SS carrier real time based on the call. Tariffs change weekly and must KCC be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I KCC can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics,
Re: [Asterisk-Users] Enhanced queue app
This might fit in with something I've worked on a bit but haven't had time to complete yet. Basically an in memory CDR modification. So the CDRs would get logged to a linked list and then try all available backends. If a backend returns an error condition then the CDR will be retried again later. So all the CDRs would be stored in memory until they have been logged to every backend available and X seconds have passed. Then there would be some applications + cli commands that would allow all that info to be pulled from the in memory tables. Once we get disconnect reasons in a standard format then that info would be available to applications as well. James On Tue, 1 Jul 2003, Jim Friedeck wrote: I think most of that information can be ascertained from the CDR database through deduction. Ideally it would be available through the management interface in realtime. Anyone feel like writing it? I don't have the time to train myself to be a Jedi-Guru Asterisk programmer and our budget is limited. I only asked for the stuff we need. Anyone else is more than welcome to try. Anyone? Bueller? Jim Friedeck -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with music during tones of dial.
On Wed, 25 Jun 2003, Xisco Mateu wrote: Inside the AGI script is call Dial application as follows: print EXEC Dial Zap/g2/number|m\n; $resultado_llamada = checkresult(); Looks like your problem lies here. The 2nd argument to Dial is the timeout. So if you don't want a timeout try: print EXEC Dial Zap/g2/number||m\n; James *CLI show application Dial -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]): ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please Help: Trying to build Asterisk - bazillionsof errors
On Sun, 22 Jun 2003, Steve wrote: Make sure you have the following installed: bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel, openssl096b, openssl-devel, readline and readline-devel. readline and readline-devel have not been needed since November of last year. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soundcore???
I assume you mean phonecore, and not soundcore. For some reason it never got added back to cvs, but I have a tarball of it before it vanished. http://asterisk.gnuinter.net/files/cvsnightly/phonecore-cvsnightly.tar.gz James On Thu, 19 Jun 2003, Roy Sigurd Karlsbakk wrote: hi all does anyone have the soundcore lib? I need it for a slackware installation of gnophone ... perhaps it's time to return it to the new cvs? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk -rx under cron?
On Thu, 19 Jun 2003, Steven Critchfield wrote: If you follow what was said above, it works interactively, but not non-interactive. Place that in crontab and it doesn't work as expected. Oops, guess I need to read more carefully. I'll look into this issue since it's likely happening in code that I wrote. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall Silly - anyone can help with a CVStar ball ?
I generate nightly tarballs of all the asterisk related sources. You can get them from http://asterisk.gnuinter.net/files/cvsnightly There are also nightly cvs changelog files at http://asterisk.gnuinter.net/files/changelogs James On Tue, 17 Jun 2003, Low, Adam wrote: Hi All, Our FW appears to be blocking my CVS attempts, does anyone have a tar ball they can send me ? Rgds, Adam * 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parking causes crash
Can you provide some more information about the problem? How are you parking the call (with #transfer, or with a hookflash on zaptel)? There was a problem with app_agent, where a segfault would occur when transfering but we fixed this late last week. If you cvs update and the problem still occurs we can try to debug this if you send the last lines of console output, a backtrace, and some more information about what you were doing James On Tue, 17 Jun 2003, John Congdon wrote: Has this been solved? When I park a call, the caller hears a second of music on hold and then the whole system crashes. I can restart with a simple (asterisk -cvvv), I don't have to reboot or anything John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] shutdown cancel?
Roy, This is already available with the 'abort halt' command James On Thu, 12 Jun 2003, Roy Sigurd Karlsbakk wrote: hi all as with the standard 'shutdown' command, it'd be nice to have a 'canceller' to 'die when convenient'. is this a heavy task to add? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ 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] FAX over IAX
I have to disagree here. I send and receive faxes over IAX all the time James On Thu, 3 Apr 2003, Brian J. Schrock wrote: From what I have heard packetizing fax does not work well, does not matter if it is IAX or SIP. I think that was straight from digium tech support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
Playback does not take an extension of the file. It looks for the best file format to use. Try using it without the .mp3 extension on there. James On Thu, 3 Apr 2003, Tamas Levente wrote: This is why I asked. (the file is there) DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-c18d, ) in new stack -- Executing Playback(SIP/levisnom-c18d, /asterisk/c.mp3) in new stack WARNING[16400]: File file.c, Line 410 (ast_openstream): File /asterisk/c.mp3 does not exist in any format WARNING[16400]: File file.c, Line 553 (ast_streamfile): Unable to open /asterisk/c.mp3 (format 4): No such file or directory WARNING[16400]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/levisnom-c18d for /asterisk/c.mp3 DEBUG[5126]: File chan_sip.c, Line 460 (__sip_ack): Stopping retransmission on '3e8ca08db129-8h0t9jtcea0u@(null)' of Response 2: Found DEBUG[5126]: File chan_sip.c, Line 721 (__sip_destroy): Detaching from SIP/levisnom-c18d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Deependencies problem
On 1 Apr 2003, Steven Critchfield wrote: On Tue, 2003-04-01 at 08:14, Ahmed Boreau wrote: hi, I'm trying to install asterisk server on mandrake 8.0 and I got a dependencies problem with libreadline.so.3 I downloaded readline-4 and readline-2.2.1 package and it still not wrking. May be some one who got this poblem could help. Did you get the dev packages? The binaries only help once a program is compiled. You need the devs to compile. I think the more troubling issue is that he must be using an extremely old version of *, since readline support was removed sometime late last year. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR ??
On Fri, 28 Mar 2003, WipeOut . wrote: Hi, I see in /ect/asterisk there is a cdr_mysql.conf to configure the CDR logging to a MySQL DB.. I have a couple of questions.. 1. Where do I find the DB schema to create the DB? (may be a good idea to add this to the top of the .conf file in the cvs so that it is easy to find for amyone wanting to set it up.. Just a thought.) The DB schema is in doc/cdr_mysql.txt but to make it easy I'll include it here: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(45) NOT NULL default '', src varchar(45) NOT NULL default '', dst varchar(45) NOT NULL default '', dcontext varchar(45) NOT NULL default '', channel varchar(45) NOT NULL default '', dstchannel varchar(45) NOT NULL default '', lastapp varchar(45) NOT NULL default '', lastdata varchar(45) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition int(11) NOT NULL default '0', amaflags int(11) NOT NULL default '0', accountcode varchar(45) NOT NULL default '' ); 2. Are there any req's to making this work? (apart from, I assume, having the mysql client installed on the * box..) I don't think you need the mysqlclient installed on the box, you just need the client libs. I haven't used the mysql cdr stuff so I can't really help here. I also read in the archives that CDR logging could be done to a CSV file, How and where is this setup and configured? by default CSV logging is enabled. You can find the logs (on a default install) in /var/log/asterisk/cdr-csv James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 100 vs SNOM 200??
On Fri, 28 Mar 2003, duncan wrote: what is WMI ? WMI didn´t work with SNOM200. i think WMI refers to Windows Management Instrumentation http://msdn.microsoft.com/library/default.asp?url=/library/en-us/dnclinic/html/scripting06112002.asp some hardware uses it for configuration (others use things like SNMP or COM). dont take my word on it though. just the only thing i could think of thats the most likely abbreviation. I have a feeling they were refering to MWI (message waiting indicator) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr showing BYEXTENSION, not actual extension
Sounds like a good idea to me. Some of the builtin help still refers to
using BYEXTENSION as well so that needs to be changed too
James
On Fri, 14 Mar 2003, Mark Spencer wrote:
Perhaps we should have BYEXTENSION print a warning that says the option is
deprecated, what do you think?
Mark
On 14 Mar 2003, Steven Critchfield wrote:
On Fri, 2003-03-14 at 10:22, Don Pobanz wrote:
We have a group of lines (FXO/FXS) between our Rolm PBX and our
Asterisk server. From the asterisk server any extension can be dialed
regardless of system. Asterisk will then either ring the appropriate *
extension or will dial a line into our Rolm PBX and dial the
appropriate Rolm extension.
The cdr works fine when an internal * phone is used. The problem is
when an outside call comes in and * answers and the caller enters an
extension which is on the Rolm PBX. What I would like to see is a call
detail records with the Rolm PBX extension that was dialed. However,
the called field of the call record contains s and the Application
and argument contain Dial and Zap/g1/BYEXTENSION.
(Zap/g1/BYEXTENSION is what is in my extensions.conf file). Instead
of seeing BYEXTENSION I would like to see the actual extension
number. Any suggestions?
Just a guess here, but try using ${EXTEN} instead of BYEXTENSION. I
think it will replace ${EXTEN} with the value before it goes to the
record.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] variable in extension.conf
On Tue, 11 Mar 2003, Tilghman Lesher wrote:
On Tuesday 11 March 2003 08:44, Steven Critchfield wrote:
On Tue, 2003-03-11 at 02:29, Rattana BIV wrote:
I try to detect if an user who use Netmeeting is connected
or not. I think in order to do that, Netmeeting-user open a
web page (in PHP) et press the button Connect or Disconnect
and the PHP set the Environnement variable which will be
proceeded in extension.conf
So i need Environnement Variable, I have test it with :
s,1,SetVar,toto=$VARENV where VARENV is my environnement
variable but toto not take the value. perhaps should I try
toto=${VARENV} or toto=${$VARENV}.
You can not pass information that way. The environment
variable for your web server is JUST for your web server. It
will not be available to asterisk. Your webserver should not
run as root, nor as the same user as asterisk. Your best bet
would be to get netmeeting to register to asterisk when it is
opened, then asterisk will know the user is available.
You might be able to write an interface in PHP to the manager
port and pass commands (perhaps to set database entries?).
Thats a good idea, currently with the manager interface you can execute
cli commands so that would be easily done. I was thinking of adding
native manager astdb commands, but I haven't had the time to get to it
recently.
James
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Re: [Asterisk-Users] astman make problems
newt is required for astman. I just looked and it looks like there is a SuSE package for newt-devel which is what you would want. I've never used SuSE so I can't help with that installation at all. The official download location for newt appears to be ftp://ftp.redhat.com/pub/redhat/code/newt but the server had too many connected users so I was unable to verify if it is in fact there. James On Mon, 10 Mar 2003, Dan Fernandez wrote: Can astman be compiled without newt? I have Suse 8.1 and it doesn´t have newt. If needed, where can I get it? Thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
