RE: [Asterisk-Users] Asterisk on Compact PCI platform
On Thu, 2004-05-20 at 01:05, Jay Milk wrote: Good call -- write cycle life of 10^3-10^4 are probably not much of an issue in a digital camera, but would probably die quickly if used as a HD replacement. Linux would have to run w/o swap. CFII+ harddrive? You're talking about a microdrive, right? While those are getting much more affordable, a laptop drive is still cheaper. The life in a digital camera isn't the concern, it is the low loss of opertunity if it fails. Not to mention it should live as long as the usefullness of the cameras. I think you are right about the cost of a laptop drive. A micro 2.2g drive is $148 plus an adapter to hook it to standard IDE. For that price you could probably get a 40gig drive in laptop form factor or 200gig in normal form factor. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Compact PCI platform
On Wed, May 19, 2004 at 05:06:09PM -0500, Steven Critchfield wrote: On Wed, 2004-05-19 at 15:12, Jay Milk wrote: Since this is related... Does anyone have Asterisk working on a Flash-drive? I was considering this as an alternative to having a harddrive in my machine, thus keeping down noise and heat. A 512MB CF card should be plenty to get Linux and * booted, another 64 or 128MB card should be plenty for voice-mail and such. Any takers? I'm sure it has been covered, but flash would not make a good long term voicemail option. Flash has a specific amount of times blocks can be written to. My company has some hand held recorders in the hands of doctors doing patient dictation, they are starting to see memory failures after a little more than a year or so of use. This is due partly because of the size of files. Each file will span multiple blocks, and therefore increases the likelyhood you will come back and write on the block again soon. A large card would stave off the problem by allowing the least used method a bit longer before reusing the blocks again. A modern CF card has about 10,000 write cycles before it starts failing. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Compact PCI platform
Hi, -Original Message- Please, don't store your dynamic sound files (voicemail) on CF. You can only write to CF so many times, so the card may start suffering failures. You can easily add a small harddrive or more memory and ramdisk software. If people really want to use CF, you could store a kernel and initrd on it and boot your system into a ramdisk. Most of the single-floppy Linux router distros work like that. This way, you're only reading from the CF once per boot (and barely writing at all). Correct. There are some scripts around on the net that do this (including commit/rollback functionality). I may have mentioned this before - one of my collegues has written a paper on this including implementation for routers etc. Can be adapted for Asterisk. Paper is dutch though :-) You would have to come up with a way to make storage of your voicemail persistent, but this could be done by using the existing feature of e-mailing the message... Or even some form of remote disk (NFS ?) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Yes, I've read and implemented all the stuff on IAX. It's the local SIP connection and its RTP streams that's the problem. For instance I noted the strange timestamp behaviour from * on local traffic earlier. Iain --On Tuesday, May 18, 2004 1:56 pm -0600 Rich Adamson [EMAIL PROTECTED] wrote: I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. The problem has been discussed multiple times over the last several weeks. To recap, there is two things needed to incure the problem: 1. cisco 7960 phone (it discards packets with uneven timestamps) 2. asterisk had an iax problem that was fixed about a month ago assoicated with uneven timestamps. The distant iax system will need to be upgraded to fairly recent code. See previous posts for more detail. ___ 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] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:42 pm -0500 Nik Martin [EMAIL PROTECTED] wrote: Out of context, this isn't much information. Is your network connection OK? Yes, AFAIK - I'm running all the traffic shaping / prioritisation stuff mentioned on the list Is your broadband provider having troubles? AFAIK - but then it is BT Openworld ;-) Has some upstream hardware changed that you may not be aware of? My call is going through IAXTEL so Digium must know if there's a problem. A test IVR system within IAXTEL would be nice for testing. Iain ___ 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] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:43 pm -0500 brian [EMAIL PROTECTED] wrote: Strange I do 7960 = * = IAX all day long without one jitter or any bad audio. Now if both ends are NOT running the very latest(within the last month or so) CVS-head for example if you have say a 2 month old chan_iax2.c on one end then oh boy you're in for a bad time they need to update. Is the 7960 using SIP? The problem happens with the latest * (cvs co asterisk). I think it's quite likely the local RTP handling that's the problem. Iain ___ 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] Softphone lag
Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards withPOTS have no such issues. Any clue where the problem lies? Thanx in advence Navnit
Re: [Asterisk-Users] AArgh, * and the 7960
I have ethereal installed and I'll do a full call trace. The Catch 22 is I
don't have access to access to a source of repeatable (ie recorded) content
accessed through IAX. That would help in producing traces for the ATA and
7960 for comparison. I mainly use IAX for non-critical international
business calls to people who wouldn't want to be * testers.
Iain
--On Tuesday, May 18, 2004 7:22 pm -0600 brian k. west [EMAIL PROTECTED]
wrote:
Lets look at this and FIX the problem instead of hacking it. What you
need to do is install etherreal and capture a call and parse the
timestamp info to see if they are slipping. Because they are perfect
here.
bkw
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 5:07 PM
Subject: Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network
connection
OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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Re: [Asterisk-Users] Softphone lag
This is normal for all VoIP communication there is nothing to wory about and the lag is not heard in normal use. Jason At 13:50 20/05/2004 +0530, you wrote: Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the problem lies? Thanx in advence Navnit
Re: [Asterisk-Users] how to pass call duration to an agi script
Hi! Ineed to pass the call duration and Bill Sec after a successfull call to an AGI script. Is there a way to do this ? - check out asterisk-addons from CVS - enable the CFLAGS+=-DMYSQL_LOGUNIQUEID in the Makefile - catch the unique ID of the call and pass it along to your AGI script - let the AGI script look up the CDR record in mySQL/postgres and then do your processing as needed Not sure if also the csv version of the CDR records does include the unique id. Cheers, Philipp ___ 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] Remote Call Forwarding
Hi! I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. The Wiki is your friend: http://www.voip-info.org/wiki-Asterisk+call+forwarding Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notify problem on sip extension
Hi! Should be mailbox = [EMAIL PROTECTED] Watch out - don't confuse an extension.conf context with a voicemail.conf context! Go to /var/spool/asterisk/voicemail and check the names of the directories (=voicemail contexts) present. Cheers, Philipp ___ 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] Fedora Core 2 and Kernel 2.6
Hi All, I decided to have a go at installing Asterisk on FC2 which now runs on Kernel 2.6.. Unfortunately I didn't get very far.. When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? Has anyone got Asterisk up and running on Kernel 2.6 yet? Later.. ___ 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] AArgh, * and the 7960
I have a 7940 running 6.3 SIP firmware and make the following type of calls:- 7940 = * = IAX2 = * = Digium X100P = Nortel CICS Analog FXS Both local and remote asterisk's run CVS-02/24/04 (built about 30mins apart). The IAX2 connection is over a VPN, and both sites are running 1500k/256k ADSL connections, about 75ms ping time between the sites. The only time I notice any problems is if one site has an application flooding its upstream, otherwise audio quality is very good. The odd packet might drop here and there, scrambling a word or two, which I usually attribute to upstream choking. 7940 is running G.729 over 100Mbps LAN to Asterisk, and IAX2 connection is presently running GSM (I've bought a couple of G729 licences for the remote asterisk but am waiting on the keys to install the beta codec). Unfortunately I don't have any spare 7940/60's at present to try out on the remote * box to see how a SIP-IAX2-SIP call would perform. ___ 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] enhanced voicemail
Hi! Should've been more clear, what I was referring to was the ability to select the first or last message as a starting point when reviewing vm's. You are referring to (4)(4) -- first msg and (6)(6) -- last msg. However I don't think this plan made it into reality. Cheers, Philipp ___ 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] disa issue
Does anyone have any experience with making DISA work? In my extensions.conf I have the line: exten = 11/2,1,DISA,|dialout The way I understood it to work is if a call comes in to 11 from 22 then if is pressed it goes to dialout, else it moves on. However, in my case, it moves directly to dialout whether is entered or nothing is entered. Can anyone enlighten me please? JC
Re: [Asterisk-Users] MGCP error dialing
Hi! I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? Do a mgcp show endpoints at the CLI and watch the output. May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01 WARNING[1251156800]: chan_mgcp.c:2608 mgcp_request: Unable to find MGCP endpoint 'aaln/[EMAIL PROTECTED]' mgcp.conf [dlinkgw] host=10.0.1.150 canreinvite=no context=default line = aaln/1 Change [dlinkgw] to [10.0.1.150], and the do a restart - depending on the Asterisk CVS version that you are using a reload or mgcp reload might not be sufficent/ might not work. See also: http://www.voip-info.org/wiki-Asterisk+config+mgcp.conf Cheers, Philipp ___ 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] Using stutter dialtone like the PSTN does
Hi! A question: is there any way to get * to answer certain DTMF sequences entered on an extension with a stutter tone? Record the stutter tone in a .wav or .gsm file and use Playback() or Background() to deliver it to the user. See also: http://www.voip-info.org/wiki-CLASS Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to pass call duration to an agi script
Philipp von Klitzing wrote: Not sure if also the csv version of the CDR records does include the unique id. It does, although not by default: cdr_csv.c: /* #define CSV_LOGUNIQUEID 1 */ /* #define CSV_LOGUSERFIELD 1 */ Apollon Koutlides ___ 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] DTMF problems to connect CME to Asterisk.
Hi.
(B
(BI want to connect Cisco7960 phone using IOS CME to Aasterisk VoiceMail system.
(BBut, DTMF relay is not work well, so I am able to hear VoiceMail intro and I am not
(Bable to contorl Asterisk VoceMail system.
(BDid anyone connect CME to Asterisk VoiceMail system?
(B
(BAsterisk machine IP is 192.168.199.250 , Cisco CME router IP is 192.168.199.254, and
(BCisco 7960 extension is 64862.
(BMy configure is like follws.
(B
(B//Asterisk sip.conf//
(B[64862]
(Btype=friend
(Binsecure=no
(Busername=64862
(Bcallerid="Cisco3600"64862
(Bhost=dynamic
(Bnat=no
(Bcanreinvite=no
(Bdefaultip=192.168.199.254
(Bdtmfmode=rfc2833
(Bmailbox=64862
(B
(B//Asterisk extensions.conf//
(Bexten = _5,2,VoicemailMain,s${CALLERIDNUM}
(B
(B//Cisco router conf//
(B!
(Bvoice service voip
(B sip
(B registrar server expires max 3600 min 360
(B!
(Bdial-peer voice 4 voip
(B destination-pattern 5+T
(B session protocol sipv2
(B session target sip-server
(B session transport udp
(B dtmf-relay rtp-nte
(B codec g711ulaw
(B no vad
(B!
(Bsip-ua
(B retry invite 3
(B retry response 3
(B retry cancel 3
(B timers trying 1000
(B registrar ipv4:192.168.199.250 expires 3600
(B sip-server ipv4:192.168.199.250
(B notify telephone-event max-duration 500
(B!
(Btelephony-service
(B load 7960-7940 P00303020214
(B max-ephones 24
(B max-dn 48
(B ip source-address 192.168.199.254 port 2000
(B timeouts interdigit 3
(B timeouts ringing 60
(B keepalive 45
(B voicemail 5#
(B!
(Bephone 1
(B keepalive 200
(B mac-address 000E.38AA.3D01
(B type 7940
(B button 1:1
(B!
(B
(BHow can I do?
(BAnd, is it possible to connect CME to Asterisk VoiceMail system?
(B
(BRegards.
(B___
(BAsterisk-Users mailing list
(B[EMAIL PROTECTED]
(Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
(BTo UNSUBSCRIBE or update options visit:
(B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI/php script not working
Dear all I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? yet if i use the agi-test.agi script everything works I don't see the difference Thanks php -q ?php fputs(STDOUT 'SAY NUMBER 123 #*\n'); $lin = fgets(STDIN); ? yet all I get on the console is -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my conf file looks like exten = 4000,1,Wait,1 ; Wait exten = 4000,2,Answer ; Answer exten = 4000,3,AGI,test.php ; run script ___ 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/php script not working
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. Iain --On Thursday, May 20, 2004 7:14 am -0400 Jer [EMAIL PROTECTED] wrote: Dear all I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? yet if i use the agi-test.agi script everything works I don't see the difference Thanks php -q ?php fputs(STDOUT 'SAY NUMBER 123 #*\n'); $lin = fgets(STDIN); ? yet all I get on the console is -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my conf file looks like exten = 4000,1,Wait,1 ; Wait exten = 4000,2,Answer ; Answer exten = 4000,3,AGI,test.php ; run script ___ 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] Asterisk on Compact PCI platform
Hi Jay, I am working on this. I am using a 256MB CF. I will keep you informed. Daniel Jay Milk wrote: Since this is related... Does anyone have Asterisk working on a Flash-drive? I was considering this as an alternative to having a harddrive in my machine, thus keeping down noise and heat. A 512MB CF card should be plenty to get Linux and * booted, another 64 or 128MB card should be plenty for voice-mail and such. Any takers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gelson Dias Santos Sent: Wednesday, May 19, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk on Compact PCI platform David H Hickman wrote: I have it working on an industrial single board pc. :) Could you post some more info about your setup? Like board brand/model, what kind of interfaces are you using and even some photos :-) Seems a very interesting project... is there anybody else running a small/compact asterisk system? I would love to have such a small system that I could send to parents, instruct them to turn it on and plug their pstn line and broadband connection and have a pstn x sip intelligent call router that requires no user intervention. Gelson David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 18, 2004, at 8:42 PM, Jacques Leisy wrote: Anybody running * on a compact PCI platform? I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000? Any hope to ever get * running on that platform? Linux Suse 9.0 is running fine Thanks Jacques ___ 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] AArgh, * and the 7960
snip But as I've mentioned before, this isn't the whole story. There are other repeatable scenarios that still cause problems, and to which some large progressive providers also see as an issue and won't accept termination becuase of it: GW - SIP - * - IAX2 - * - SIP - 79X0 Now, if this scenario has been corrected as well, please accept my apologies for bringing it up. This config, with the absolute latest CVS HEAD, well as of a week or so ago when I last checked, seems to cause issues on the sequencing. I seem to recall comments that there is some work still being done on getting this cross protocol packet sequencing to work properly? I'll have to get Ethereal out again and prove that it is still happening. And why are we blaming Cisco for dropping packets that are mis-sequenced, when we shouldn't be sending them mis-sequenced packets in the first place? Assuming there is a sequence numbering issue (which I don't doubt, I just haven't taken the time to investigate), no one is blaming cisco. Rather, the sequence problem would be another issue. Cisco's problem is that it drops packets (choppy audio) when the rtp timestamps within the rtp pkt are not consistent. Totally unrelated to sequence numbers. ___ 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/php script not working
Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ 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: Fedora Core 2 and Kernel 2.6
Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz hyperthreading CPU 1G RAM. I decided to use kernel 2.6 after reading about problems with hyperthreading and asterisk in 2.4 on this list. So far I've only connected to VOIP service providers and everything has been working very well. I will however connect a PRI line in the next 3-4 weeks so I'm interested in hearing from experienced kernel 2.6 users as well. I'm also interested in getting in contact with people using asterisk as a hotel pbx, which is my setup (100 rooms in 3 locations, 1 asterisk box). Best regards, Maron Kristofersson WipeOut wrote: Hi All, I decided to have a go at installing Asterisk on FC2 which now runs on Kernel 2.6.. Unfortunately I didn't get very far.. When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? Has anyone got Asterisk up and running on Kernel 2.6 yet? Later.. ___ 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] Problems with Quadbri card
Hello, We have been running a system based around the quadbri card from www.junghanns.net for around 3 weeks now. For the first two weeks everything was stable and ran well. In the last week a issue has appeared, described below: Someone attempts to call us, * sees an incoming call (it is anounced on the console e.g. Accepting call from '' to '781950' on channel 2, span 2) and * says it has picked up the call. However the person calling isn't connected, hears no ringing and gets a tone as if the call has been rejected. This happens on some but not all calls. As * believes the call is connected the channel remains open and the call remains in the system until it is manually cleared. It doesn't appear to be a service provider issue, they;ve tested the lines and see no problem. The only strange error message in the logs is: WARNING[15376]: PRI: !! Don't know what to do with M3=7 u-frames which has started to appear recently bu it occurs once to twice a day where as the failed calls can happen tens of times a day. We have 2 isdn2e line with 1 hunt number. We were on bristuff 0.0.2rc20a and have changed to the latest 0.0.2 but the problem still occurs. Any idea on how to resolve this would be greatly appreciated. regards, Julien Levi ___ 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] Problems with Quadbri card
Also, since installing 0.0.2 we see this occasionally in the logs: May 19 18:59:26 WARNING[16400]: Ring requested on channel 1 already in use on span 1. Hanging up owner. May 19 18:59:32 WARNING[16400]: Ring requested on channel 2 already in use on span 1. Hanging up owner. May 19 19:00:10 WARNING[16400]: Call specified, but not found? May 19 19:00:26 WARNING[16400]: Call specified, but not found? ___ 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] AArgh, * and the 7960
On Thu, 20 May 2004, Iain Stevenson wrote: The Catch 22 is I don't have access to access to a source of repeatable (ie recorded) content accessed through IAX. That would help in producing traces for the ATA and 7960 for comparison. The payload (i.e. audio) of the RTP stream is not relevant, at least in my experience. All the information you need is in the RTP header -- sequence numbers (not a problem, that I've seen) and timestamps. If you have two SIP phones, a FWD account and an IAXtel account, you have all you need to test SIP-IAX2-SIP. From one SIP phone, use your FWD account to call your IAXtel number, and pick up the incoming call on your other SIP phone. To avoid looping issues (multiple hops through your * box), make the source (FWD) end a SIP client defined directly to FWD, the IAXtel end your * box, and hang your destination SIP client off *. Subject to the bandwidth you have available upstream, this should be an adequate test and allow you to capture everything you need. Capture everything in and out of the * box if you can, as this will give the greatest amount of information and good correlation between the IAX2 traffic and the SIP traffic that goes to your SIP destination. Hope this is helpful (and not restating the bleeding obvious)... Cheers, Vic Cross ___ 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] MultiTech MVP200 and Iconnect
Hi, I try to use IConnect on my MultiTech MVP 200 VoIP Gateway and didnt workL. I try thru my asterisk box and everything works fine The MVP200 is behind the Nat and my * is connected directly on Internet exactly like IConnect. Thanks in advance for any help. Chris HARIGA smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] Call recording between SIP phones
I'm currently doing it successfully using the Monitor command. There's a really good example on the Wiki (www.voipinfo.org) (just search for 'Monitor'). After following the Wiki example, I then use the SetCDRUserField to store the recording filename in the CDR (using MySQL) so I can display it with a simple PHP page so my users can search for and listen to their recorded conversations. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hollinger Sent: Wednesday, May 19, 2004 4:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call recording between SIP phones It does work but you need to add canreinvite=no to your sip.conf to keep asterisk in the audio loop. Hi everybody, I have been searching around for days on how to record calls between SIP phones.Could someone tell me whether it is possible? The Record command doesn't seem to work during a call. Thanks Lamine ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail customization
have managed to establish voicemail functionality using voicemail / voicemailmain applications the documentation on these applications from digium.com suggests that voicemail greetings are customizable (as one would be expect), but am not able to find any supporting documentation can anyone refer me to said documentation or provide assistance on how to proceed GT ___ 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/php script not working
OK, but I have AGI working and you don't - so please allow me the error since it's a while since I worked on this, Of course, it would help if * used consistent syntax for identical commands in extensions.conf and AGI, but that's another debate. Why not check the logs for php and * and post anything relevant here. Enable the maximum debugging support in *. Iain --On Thursday, May 20, 2004 2:44 pm +0300 Apollon Koutlides [EMAIL PROTECTED] wrote: Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail customization
Hi! have managed to establish voicemail functionality using voicemail / voicemailmain applications the documentation on these applications from digium.com suggests that voicemail greetings are customizable (as one would be expect), but am not able to find any supporting documentation can anyone refer me to said documentation or provide assistance on how to proceed The Wiki is your friend, even on a sunny day: http://www.voip-info.org/wiki-Asterisk+VoiceMail Cheers, Philipp ___ 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] RADIUS acc module for Asterisk
does any one has that can give it to me RADIUS acc module for Asterisk. Im realy interested on an account an billing manager. Best regards Hekuran Doli ___ 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] AArgh, * and the 7960
The Catch 22 is I don't have access to access to a source of repeatable (ie recorded) content accessed through IAX. That would help in producing traces for the ATA and 7960 for comparison. The payload (i.e. audio) of the RTP stream is not relevant, at least in my experience. All the information you need is in the RTP header -- sequence numbers (not a problem, that I've seen) and timestamps. If you have two SIP phones, a FWD account and an IAXtel account, you have all you need to test SIP-IAX2-SIP. From one SIP phone, use your FWD account to call your IAXtel number, and pick up the incoming call on your other SIP phone. To avoid looping issues (multiple hops through your * box), make the source (FWD) end a SIP client defined directly to FWD, the IAXtel end your * box, and hang your destination SIP client off *. Subject to the bandwidth you have available upstream, this should be an adequate test and allow you to capture everything you need. Capture everything in and out of the * box if you can, as this will give the greatest amount of information and good correlation between the IAX2 traffic and the SIP traffic that goes to your SIP destination. I might add to Vic's comments that simply signing up for an FWD IAX account is enough for testing in most cases. They provide a consistent source of audio in the forms of a milliwatt generator, data-time annoucements, and other automated sources of audio to generate the rtp stream. Some of those sources may have other issues, but they are sufficiently stable to observe sequence numbers, timestamps, etc. It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI/php script not working
Hi! I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? Look here: http://www.voip-info.org/wiki-Asterisk+AGI+php php -q ?php fputs(STDOUT 'SAY NUMBER 123 #*\n'); $lin = fgets(STDIN); ? Is that really your entire script, or did you cut out most of the lines? If not, then I am afraid that you need to do quite some reading on both AGI and PHP first. Cheers, Philipp ___ 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/php script not working
At 08:47 AM 5/20/2004, you wrote: -vvvc mode I see *CLI -- Executing Wait(Phone/phone0, 1) in new stack -- Executing Answer(Phone/phone0, ) in new stack -- Executing AGI(Phone/phone0, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my question is why don't I see the output on stderr OK, but I have AGI working and you don't - so please allow me the error since it's a while since I worked on this, Of course, it would help if * used consistent syntax for identical commands in extensions.conf and AGI, but that's another debate. Why not check the logs for php and * and post anything relevant here. Enable the maximum debugging support in *. Iain --On Thursday, May 20, 2004 2:44 pm +0300 Apollon Koutlides [EMAIL PROTECTED] wrote: Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ 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] AArgh, * and the 7960
It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ 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] Softphone Audio problem
As a test, I was trying to use Iaxcomm and Iaxphone to connect to Asterisk and dial out to my other line. Using either of these soft phones, I can connect to Asterisk and listed to audio just fine. I can even connect across the net to another asterisk server and hear audio just fine, however, when I dial out to my second land line the audio that is transmitted is horribly broken up. It is as if the audio stream is broken into 8 parts every second and then every other part is dropped. I then tried the asterisk echo test and got the same thing. I am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the soft phones on my laptop running Windows XP (Laptop is Sony Vaio PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). Is this an asterisk problem or a soft phone problem? If asterisk, any ideas on how to fix it? Thanks, Andy Farnsworth ___ 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] Grandstream tftp cfg.txt format
Hello! I've been reading through the archives on this list for the last 8-10 months. There are some reports on success with tftp autoconfiguration with a given cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that creates such a file, but I need the format to do that. Regards, Maron Kristofersson ___ 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] AArgh, * and the 7960
www.bkw.org/~web/parse.txt That should parse and show ALL lines where the timestamps slip. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Thursday, May 20, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ 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] Re: Asterisk on Compact PCI platform
Good call -- write cycle life of 10^3-10^4 are probably not much of an issue in a digital camera, but would probably die quickly if used as a HD replacement. i have a cigarette-sized freebad box using only flash for disk and swap. runs for years under load. a bunch of us use them, though the rest of the folk are netbsd deviants:-) . randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
http://www.bkw.org/~brian/parse.txt Its still early. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Thursday, May 20, 2004 8:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] AArgh, * and the 7960 www.bkw.org/~web/parse.txt That should parse and show ALL lines where the timestamps slip. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Thursday, May 20, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as channel 1 and
zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then issues the
following errors:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at this time
My config files are below:
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G723.1
context=from-sip
[2000]
type=friend
username=2000
secret=
host=dynamic
mailbox=2000
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
PSTN-1=Zap/1
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
include = to-pstn
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
Can anyone help me out here?
Thanks,
Scott
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[Asterisk-Users] Tellabs 2572 Configuration Advice?
Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find anywhere. I've registered for the Tellabs portal both recently and in the past 6 months, but I never heard back from them. Any help at all would be appreciated! Regards, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p card + dailing out
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at this time
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
You're missing a $ on the second dial line on {PSTN-1}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] x100p card + dailing out
Thanks for the syntax error fix, but I'm still having the same problem.
Funny thing was, I never caught that syntax error because so far I was only
trying with the preceding '1'.
I can't seem to find this error relating to the x100p cards via google, the
asterisk mailing list archives, or the wiki.
Any other ideas?
Scott
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 10:59 AM
Subject: Re: [Asterisk-Users] x100p card + dailing out
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at this time
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
You're missing a $ on the second dial line on {PSTN-1}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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[Asterisk-Users] Mysql
Hi, to all!!! I can't download asterisk-addons...I try with CVS, but i can't. How can I do??? Thank you Fabio
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. -- Joshua M. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FC2 compile of zaptel
I just grabbed my fresh Fedora Core 2 final release. Untared zaptel-0.9.1 dir make linux26 and I get errors on the compile. Anyone else tried this yet and been sucessful? I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6 But I still get errors after that... about asm/linkage.h asm/types.h -- Jerry Geis MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 (240)282-0319 Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budgetone problem on hangup
Hello to all.
I have a couple of budgetones connected to Asterisk
server. I can establish calls using budgetone with no
problem, but when I hang up a Budgetone, Asterisk
does not detect the hangup. It seems that the
communication goes on in spite of budgetone's hangup.
My sip.conf:
[general]
disallow=all
allow=ulaw
bindaddr=172.16.60.21
[sip1]
callgroup=1
pickupgroup=1
type=friend
secret=sip1
auth=md5
host=dynamic
reinvite=no
canreinvite=no
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
callerid=sip1 101
context=telefonos
[sip2]
callgroup=1
pickupgroup=1
type=friend
secret=sip2
auth=md5
host=dynamic
reinvite=no
canreinvite=no
callgroup=1
pickupgroup=1
language=es
dtmfmode=rfc2833
callerid=sip2 102
context=telefonos
extensions.conf:
[globals]
EXTEN106=Sip/sip1
EXTEN107=Sip/sip2
[telefonos]
exten = _1XX,1,NoOp(${CALLERID})
exten = _1XX,2,Dial(${EXTEN${EXTEN}},,tT)
exten = _1XX,3,Hangup
exten = _1XX,103,VoiceMail2(u101)
_
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Información de Estados Unidos y América Latina, en Yahoo! Noticias.
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[Asterisk-Users] Mystery SIP channels
Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ 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] FC2 compile of zaptel
Jerry Geis wrote: I just grabbed my fresh Fedora Core 2 final release. Untared zaptel-0.9.1 dir make linux26 and I get errors on the compile. Anyone else tried this yet and been sucessful? I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6 But I still get errors after that... about asm/linkage.h asm/types.h See post above by Joshua M Thompson. Same issue? ___ 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] x100p card + dailing out
Post your zapata.conf and zaptel.conf
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
Sent: Thursday, May 20, 2004 9:45 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as
channel 1 and zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then
issues the following errors:
channel.c:1676 ast_request: No channel type registered for
'{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel
of type '{PSTN-1}' = = Everyone is busy at this time
My config files are below:
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G723.1
context=from-sip
[2000]
type=friend
username=2000
secret=
host=dynamic
mailbox=2000
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
PSTN-1=Zap/1
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
include = to-pstn
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
Can anyone help me out here?
Thanks,
Scott
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RE: [Asterisk-Users] Mystery SIP channels
What address is that? Is it a phone (or address of a PC with a softphone?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, May 20, 2004 10:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mystery SIP channels Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ 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] Strange Sip (FWD, SipGate and such) problem
Karl Brose wrote: I think when you have this setup you need to keep the media path going through Asterisk at all times. Your SIP is binding to both ports, internal and external, but that doesn't correctly set it up for either scenario, localnet calls and external calls. It won't keep the addresses straight for the RTP channels. Try setting CANREINVITE=NO for peers (FWD,..) and for your local SIP phones. When a channel is created in asterisk the media path is going through Asterisk, but during a call the endpoints can issue reinvites which switches the media path directly between the endpoints. You need to prevent that. Other solutions are to run IAX to/from FWD and SIP locally, or SIP to the external peers and IAX to a local IAX phone (or another protocol). Or you should be able to create your own NAT using the iptables and bind asterisk only on one port either outside or inside and set the right corresponding parameters. The RTP will still bind on all ports currently, but that will be fixed in a matter of days. Also, sipgate.net should be sipgate.de (works ok though since they don't care) fromdomain is meant to be realm not a hostname. Thomas Gallaway wrote: Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp(SIP/113-6d2e, ) in new stack -- Executing Goto(SIP/113-6d2e, intern-post|714551|1) in new stack -- Goto (intern-post,714551,1) -- Executing SetCallerID(SIP/113-6d2e, 270002) in new stack -- Executing SetCIDName(SIP/113-6d2e, Thomas Gallaway) in new stack -- Executing Dial(SIP/113-6d2e, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd270002-6ee7 answered SIP/113-6d2e -- Attempting native bridge of SIP/113-6d2e and SIP/fwd270002-6ee7 == Spawn extension (intern-post, 714551, 3) exited non-zero on 'SIP/113-6d2e' But either I get 1/2 second of audio or no audio. No matter how long I wait there is just no audio or just a short snippet of audio at the beginning. Here is parts of my sip.conf; [general] port = 5060 ; Port to bind to localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask externip = 206.40.161.235 context = intern; Default for incoming calls maxexpirey=3600 defaultexpirey=300 disallow=all; Disallow all codecsa allow=gsm allow=alaw allow=ulaw tos=reliability register = xxx:[EMAIL PROTECTED]/150 register = xxx:[EMAIL PROTECTED]/151 [sipgate1] type=friend username=xxx secret=xxx host=sipgate.de fromuser=xxx fromdomain=sipgate.net nat=no context=incoming-sipgate canreinvite=yes [fwd270002] allow=ulaw type=friend context=incoming-fwd secret=xxx username=xxx host=fwd.pulver.com Any ideas? When I put nat=yes I actually will get 1 second of audio, then it dies. I have been googling for a while now and not seem to find any sollution to this. -- Thomas I will try that. I had to remove all the IAX / SIP changes I did as even on the local network it started to give me a one way communication thing. I was able to hear other people but they could not hear me. Will give this another try later in the week. -- Thomas ___ 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] Mystery SIP channels
I had the same thing come up on mine when I was having codec issues with one of my phones. Kyle Nik Martin wrote: What address is that? Is it a phone (or address of a PC with a softphone?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, May 20, 2004 10:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mystery SIP channels Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Premisys Slimline CB
I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. 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] G729 codec for asterisk
Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. I am really in a hurry here. Please answer as soon as possible. Pablo Salinas RD - CONEXION S.A. Asuncion - Paraguay (S. America) Phone:595-21-440104 Fax: 595-21-440270 ___ 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] x100p card + dailing out
You might want to try removing the hyphen. It could be misinterpreting
it? Might want to try simplifying things a bit too for testing
purposes. Take out the PSTN-1 and put in the ZAP/1 directly into your
dial plan to verify that * can access the ZAP channel correctly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
Sent: Thursday, May 20, 2004 10:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] x100p card + dailing out
Thanks for the syntax error fix, but I'm still having the same problem.
Funny thing was, I never caught that syntax error because so far I was
only trying with the preceding '1'.
I can't seem to find this error relating to the x100p cards via google,
the asterisk mailing list archives, or the wiki.
Any other ideas?
Scott
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 10:59 AM
Subject: Re: [Asterisk-Users] x100p card + dailing out
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:
channel.c:1676 ast_request: No channel type registered for
'{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of
type '{PSTN-1}' = = Everyone is busy at this time [to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
You're missing a $ on the second dial line on {PSTN-1}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Softphone lag
I always get that with a softphone, but not with a hardphone. Grandstream BT100 is only $70 so Im gonna get those for most of the people. And a few higher end phones for the execs. Kyle Navnit Chachan wrote: Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the problem lies? Thanx in advence Navnit ___ 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] Fedora Core 2 and Kernel 2.6
Joshua M. Thompson wrote: On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. Thanks for the try but its didn't work.. Got exactly the same result.. Anything else I can try? ___ 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] Mystery SIP channels
I don't actually know. All of the users are behind NAT, so the channel list doesn't match the peers list. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nik Martin Sent: Thursday, May 20, 2004 10:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mystery SIP channels What address is that? Is it a phone (or address of a PC with a softphone?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, May 20, 2004 10:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mystery SIP channels Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102(None) df92fb1b-8a 00101/03059 0ms ms UNKN ___ 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] G729 codec for asterisk
Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. No you MUST pay per channel because the patent holders require that. The patent holders would [EMAIL PROTECTED] kittens if you had no port limit or any type of control on it. That's why the control and registration processes are in place to comply with the patent holders requirements. So your request translates into I want something for nothing. bkw ___ 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] G729 codec for asterisk
If you are in a hurry then you should call Digium On Thu, 2004-05-20 at 10:58, pesb wrote: Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. I am really in a hurry here. Please answer as soon as possible. Pablo Salinas RD - CONEXION S.A. Asuncion - Paraguay (S. America) Phone:595-21-440104 Fax: 595-21-440270 ___ 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 * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Time Limit Warning File
Hi, Im playing with the CVS head time limiting at Dial application, it just works fine but the only problem is that the caller isnt hearing the warning message. Im using a Cisco 7960 as the caller and a Polycom 500 as the callee. The audio is passing through Asterisk: -- Executing Dial(SIP/8992-9712, SIP/8988|20|L(1:2000)) in new stack -- Limit Data: -- timelimit=1 -- play_warning=2000 -- play_to_caller=yes -- play_to_callee=no -- warning_freq=0 -- start_sound=UNDEF -- warning_sound=timeleft -- end_sound=UNDEF -- Called 8988 -- SIP/8988-6922 is ringing -- SIP/8988-6922 answered SIP/8992-9712 == Spawn extension (local, 8988, 1) exited non-zero on 'SIP/8992-9712' If I change the LIMIT_WARNING_FILE to something like 'beep' to use the usual beep.gsm file same results :( Any suggestions ? -- Juanjo sin .sig ___ 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] x100p card + dailing out
Here you go:
zaptel.conf
fxsks=1
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
echocancel=yes
echocancelwhenbridged=yes
context=from-pstn
signalling=fxs_ks
callerid=asreceived
channel=1
Scott
- Original Message -
From: Nik Martin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 11:45 AM
Subject: RE: [Asterisk-Users] x100p card + dailing out
Post your zapata.conf and zaptel.conf
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
Sent: Thursday, May 20, 2004 9:45 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as
channel 1 and zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then
issues the following errors:
channel.c:1676 ast_request: No channel type registered for
'{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel
of type '{PSTN-1}' = = Everyone is busy at this time
My config files are below:
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G723.1
context=from-sip
[2000]
type=friend
username=2000
secret=
host=dynamic
mailbox=2000
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
PSTN-1=Zap/1
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
include = to-pstn
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
Can anyone help me out here?
Thanks,
Scott
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Re: [Asterisk-Users] Terrible TICKING sound
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anon wrote: | On Thursday 13 May 2004 11:57 pm, Jason A. Pattie wrote: | |-BEGIN PGP SIGNED MESSAGE- |Hash: SHA1 | |Steven Critchfield wrote: || On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote: ||Our problem ended up not being with Asterisk or Digium hardware. It was ||the analog cordless phone. We simply have to live with it. What ||happens is whenever a connection is established and the phone is ||off-hook, an LED on the base lights up in a blink blink . blink ||blink . etc. pattern. Everytime the LED lights, a pulse is sent to ||the phone. It's especially bad when both lines are in use, as the phone ||is a two-line capable device. Then you've got double the pulsing. || ||This may have nothing to do with your problem. Just wanted to get it ||out there in case anyone else runs into it, too. || || Sounds like your phone needs either a aux power source to power that || led, or possible a little modification to clip that LED. || || I would make sure your cordless phone's power supply is within spec. If || it is, Maybe you might want to look into one of the other comments a || while back on the list about upping the power on the SLIC(?). You might || be able to provide enough power to the phone to not cause trouble when || it blinks the LED. | |Well, the phone is using the power supply that came in the box. :) | | If the phone is old and had average or more use, the transformer in the | wall-wart might be operating at less capacity than when it was new, and | might not be adequate now. Hmm. It's fairly new. Less than 2 years old. | You can get a good, inexpensive replacement wall-wart from www.jameco.com. Neat website. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFArNt0uYsUrHkpYtARAjc+AJ9mXFPM67A9LTNED4Bsr1GR21KWrACfeEqc oCLJCOWgyjP8mfk3nsbTCoU= =oBpH -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 codec for asterisk
This is discussed at length in the Wiki, on several pages, including: http://www.voip-info.org/wiki-Asterisk+G.729+licensing Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of pesb Sent: Thursday, May 20, 2004 8:59 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 codec for asterisk Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. I am really in a hurry here. Please answer as soon as possible. Pablo Salinas RD - CONEXION S.A. Asuncion - Paraguay (S. America) Phone:595-21-440104 Fax: 595-21-440270 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone lag
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jason Williams wrote: | This is normal for all VoIP communication there is nothing to wory about | and the lag is not heard in normal use. | | Jason | | At 13:50 20/05/2004 +0530, you wrote: | | Hi, | IF i use a sip softphone or a iax softphone with asterisk, i get a lag | of about 1 second. | The two phones were on 2 different pc's near me. When I speak on one, | i hear it on the other after about 1 second. | I tried using iaxComm, Xten Xlite, etc. Same. | | FYI: The codec used was GSM. Would it get better if a straight codec were used? I.e., one that does full 8KBps (I think that's either ALAW or ULAW). In other words, bandwidth usage would obviously increase over a codec like GSM, but would the lag be reduced because no translation has to be done at all, just send it to the audio device, no (de)compression, etc.? - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFArNyauYsUrHkpYtARAu3xAJ9Fs0DBy84FrKvNRFVJ33xKk44EmwCfakDp IJTNikr6bb04fAJyYaiW0Ak= =HphP -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
WipeOut wrote: Joshua M. Thompson wrote: On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. Thanks for the try but its didn't work.. Got exactly the same result.. Anything else I can try? It really sounds like your do not have the kernel-headers installed. I never tried a 2.6 kernel but on 2.4 I got similar errors until I installed the kernel-headers. How did you get the kernel header files for FC2? ___ 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] Free Softphone Recomendations
Hi, - Original Message - From: Tor Houghton You can enable the key beep in DIAX, but what's the reason to get a DTMF type of feedback? The beep is not enough? For some people, maybe? I just find it more natural to hear the DTMF when I hit a number. It means that if I am dialling a number, I get aural feedback of what I've pressed, which means that I can hear whether or not I've dialled the wrong numbers (e.g. while I am not looking at the screen). I will put this on the wish list. :-) Dan P.S. You can really decode DTMF tones with your ear/brain?..:-) ___ 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] G729 codec for asterisk
Yes this is correct, you need too purchase licenses, but the number of licenses you buy mast be proportional to the size of the cpu's processor you have. Jorge Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. I am really in a hurry here. Please answer as soon as possible. Pablo Salinas RD - CONEXION S.A. Asuncion - Paraguay (S. America) Phone:595-21-440104 Fax: 595-21-440270 ___ 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] Fedora Core 2 and Kernel 2.6
On Thursday 20 May 2004 11:04, WipeOut wrote: Joshua M. Thompson wrote: You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. Thanks for the try but its didn't work.. Got exactly the same result.. Anything else I can try? make -C /usr/src/linux dep -- 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
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
On Thu, 2004-05-20 at 12:04, WipeOut wrote: Thanks for the try but its didn't work.. Got exactly the same result.. Apparently the FC2 2.6.5 kernel has another issue, one that I didn't start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few files that are auto generated by the Makefile. The complete directions to set up your source tree are thus: cp configs/config-for-my-kernel .config make oldconfig make include/asm make include/linux/version.h make SUBDIRS=scripts A pain in the butt but at least you only have to do this once after installing a new kernel-source RPM. -- Joshua M. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse broken?
Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ 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] Tellabs 2572 Configuration Advice?
Jeff Noxon wrote: Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find anywhere. I've registered for the Tellabs portal both recently and in the past 6 months, but I never heard back from them. Any help at all would be appreciated! Regards, Jeff I'm using a 2571 between a T100P and Premisys CB with really good success. I could never seem to master the push buttons on the front of the card. The menu-driven, serial hookup in the back of the card cage is much superior. I use mincom set for 9600 baud, 7 bit, even parity. Type @1 to get the attention of the first card. Also, all commands must be in upper case. The menus pretty much guide you through the setup. If you still have problems, I'll send you some screen snaps of my config. Good luck, Steve ___ 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] asterisk-providers mailing list?
It seems like it might be nice to have a mailing list to talk about (and to) voip providers for Asterisk users. It would be a good place to share info about config, pricing news, customer service, local numbers, transient outages, etc. Providers would be encouraged to contribute sales info. Users would be able to help each other out with technical and non-technical issues. Seems good for everyone and it would keep some of the noise and hurt feelings out of the other lists. The real goal of the list would be to improve the quality of the experience for customers and suppliers. This is something we need to improve in order for voip to be taken more seriously. ? -reed ___ 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] Error running festival command
I'm finding I can't run two festival commands in the same connection. Given the following: exten = 555,1,Answer exten = 555,2,Wait(1) exten = 555,3,Festival(mary had a little lamb) exten = 555,4,Wait(1) exten = 555,5,Festival(she also had a duck) exten = 555,6,Hangup Calling 555 gets the first line, then I get the error: May 20 17:59:16 WARNING[1301883824]: rtp.c:386 ast_rtp_read: RTP Read error: Resource temporarily unavailable ..and the line goes dead (but stays active). I guess I could just record the output from festival and stuff it in the sounds directory, but it seems a hack rather than a solution. Would upgrading to the CVS version help? I haven't gone live with the system yet (still waiting for the FXO card to be shipped from the US) so am fairly flexible at the moment... Tony ___ 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] x100p card + dailing out
I removed the PSTN-1 variable reference and started referencing it as Zap/1
and also ZAP/1, without any difference - same errors.
I believe the hyphen you were talking about was the one in PSTN-1.
Scott
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 12:01 PM
Subject: RE: [Asterisk-Users] x100p card + dailing out
You might want to try removing the hyphen. It could be misinterpreting
it? Might want to try simplifying things a bit too for testing
purposes. Take out the PSTN-1 and put in the ZAP/1 directly into your
dial plan to verify that * can access the ZAP channel correctly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
Sent: Thursday, May 20, 2004 10:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] x100p card + dailing out
Thanks for the syntax error fix, but I'm still having the same problem.
Funny thing was, I never caught that syntax error because so far I was
only trying with the preceding '1'.
I can't seem to find this error relating to the x100p cards via google,
the asterisk mailing list archives, or the wiki.
Any other ideas?
Scott
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 10:59 AM
Subject: Re: [Asterisk-Users] x100p card + dailing out
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:
channel.c:1676 ast_request: No channel type registered for
'{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of
type '{PSTN-1}' = = Everyone is busy at this time [to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten = _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
You're missing a $ on the second dial line on {PSTN-1}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
Joshua M. Thompson wrote: On Thu, 2004-05-20 at 12:04, WipeOut wrote: Thanks for the try but its didn't work.. Got exactly the same result.. Apparently the FC2 2.6.5 kernel has another issue, one that I didn't start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few files that are auto generated by the Makefile. The complete directions to set up your source tree are thus: cp configs/config-for-my-kernel .config make oldconfig make include/asm make include/linux/version.h make SUBDIRS=scripts A pain in the butt but at least you only have to do this once after installing a new kernel-source RPM. Well done Joshua!!.. I have no idea what all that just did but it looks like Zaptel has built.. I won't be able to test the drivers for a while with and actual card but at least i can now try build Libpri and Asterisk.. Thanks.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone Audio problem
On Thu, 20 May 2004 14:18:16 +0100, Andy Farnsworth [EMAIL PROTECTED] wrote: As a test, I was trying to use Iaxcomm and Iaxphone to connect to Asterisk and dial out to my other line. Using either of these soft phones, I can connect to Asterisk and listed to audio just fine. I can even connect across the net to another asterisk server and hear audio just fine, however, when I dial out to my second land line the audio that is transmitted is horribly broken up. It is as if the audio stream is broken into 8 parts every second and then every other part is dropped. I then tried the asterisk echo test and got the same thing. I am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the soft phones on my laptop running Windows XP (Laptop is Sony Vaio PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). Is this an asterisk problem or a soft phone problem? If asterisk, any ideas on how to fix it? What kind of PSTN interfaces are you using? I'm not sure from the description: are you seeing the problem of both lines, or only the second line? Do you get the same kind of results when using a SIP softphone? I'll be posting new binaries to sourceforge this weekend, because there have been some library changes related to jitter, but I haven't heard or seen anything as drastic as you describe. BTW what version of asterisk? Thanks, Andy Farnsworth ___ 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] snom 200 and hold
Hi, I've looked through the archives and seen references to placing calls on hold on a snom 200 (any version of the firmware but we have the latest: 2.05e.) Basically, we can't place calls on hold on the snom 200! The manual talks about the Flash button (which is really the R button, as far as I can tell.) Pressing the R button will immediately disconnect the incoming call. Another poster to this list indicated one could just choose another line and the current line will be put on hold. This is not true on our phone: again, the original call is immediately disconnected. We've been all over the settings in the snom 200 and have tweaked a bunch of parameters. So: how does one place an incoming call on hold on a snom 200 so that we can do attended transfer? Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anonymous sip register
Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? Im thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad
Re: [Asterisk-Users] Re: Fedora Core 2 and Kernel 2.6
Maron Kristófersson wrote: Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz hyperthreading CPU 1G RAM. I decided to use kernel 2.6 after reading about problems with hyperthreading and asterisk in 2.4 on this list. So far I've only connected to VOIP service providers and everything has been working very well. I will however connect a PRI line in the next 3-4 weeks so I'm interested in hearing from experienced kernel 2.6 users as well. I'm also interested in getting in contact with people using asterisk as a hotel pbx, which is my setup (100 rooms in 3 locations, 1 asterisk box). If you hit a wall trying to get intel based boxes to do the job, let me know. I am working on a SunOS port. It would be fun to see this running on a Sun Fire server. Should be able to scale it to 1000+ rooms. Only problem, servers run from about 50k to a million. That's like real money. But it would still be fun. btw: this is not a very pretty port. The current state of the * source tree does not lend itself very well to other OS's. Quite a bit of hacking involved. Something that I would never want to see checked into cvs. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] VoicePulse broken?
I am having an issue with voicepulse also. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Sullivan Sent: Thursday, May 20, 2004 12:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ 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] asterisk-providers mailing list?
I thought that is what the Asterisk-Biz mailing list was for. http://lists.digium.com/mailman/listinfo/asterisk-biz On Thu, 2004-05-20 at 12:04, Reed Wade wrote: It seems like it might be nice to have a mailing list to talk about (and to) voip providers for Asterisk users. It would be a good place to share info about config, pricing news, customer service, local numbers, transient outages, etc. Providers would be encouraged to contribute sales info. Users would be able to help each other out with technical and non-technical issues. Seems good for everyone and it would keep some of the noise and hurt feelings out of the other lists. The real goal of the list would be to improve the quality of the experience for customers and suppliers. This is something we need to improve in order for voip to be taken more seriously. ? -reed ___ 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 * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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:Remote Call Forwarding
Philipp, I already have that call-forwarding feature set into asterisk. What I am looking is how to set that feature remotely by calling into your voicemail or any given no. so that person can set call-forwarding remotely. Few of our sales people want this kind of feature, because if they are stuck in traffic and expecting important call, so that, they can call from there mobile into asterisk and set call-forward to there mobile. With the current call-forwarding feature, person has to be there physically to set this feature from there extension. If somebody has any example, it would be great help. Regards, KD Date: Thu, 20 May 2004 11:02:31 +0200 From: Philipp von Klitzing [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Remote Call Forwarding To: [EMAIL PROTECTED] Organization: AEGEE Reply-To: [EMAIL PROTECTED] Hi! I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. The Wiki is your friend: http://www.voip-info.org/wiki-Asterisk+call+forwarding Cheers, Philipp ___ 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-providers mailing list?
At 1:04 PM -0400 on 5/20/04, Reed Wade wrote: It seems like it might be nice to have a mailing list to talk about (and to) voip providers for Asterisk users. It would be a good place to share info about config, pricing news, customer service, local numbers, transient outages, etc. Providers would be encouraged to contribute sales info. Users would be able to help each other out with technical and non-technical issues. Seems good for everyone and it would keep some of the noise and hurt feelings out of the other lists. The real goal of the list would be to improve the quality of the experience for customers and suppliers. This is something we need to improve in order for voip to be taken more seriously. ? -reed Would providers actually contribute meaningful discussion and data on such a list? My experience shows that the majority of providers that I know (and have worked with or for) and who use Asterisk have not once, ever, posted anything to either the -dev list or the -users list. That number is more than ten and less than thirty, to be suitably vague. In fact, the only activity on any VoIP list or organizations from any of the providers I've worked for seems to be... me. This is not to say that I'm always the only VoIP person at these firms (though that has certainly been the case at several) but it does say that providers are notoriously secretive and closed-mouth, and automatically distrustful of anything that could expose them to the shame of running software or hardware that didn't cost them millions of dollars. (frantic_hand_waving We're HUGE! We're MAMMOTH! YOU SHOULD INVEST IN US! We've spent INCREDIBLE AMOUNTS OF MONEY on this system to bring AMAZING RESULTS to our customers and INVESTORS! It's IMPOSSIBLE to duplicate what we've done! Ar!) I would love to see a list where this speak-no-evil trend is reversed, but I suspect it would be a very low-volume list. There is already a list called isp-clec which (sometimes) covers this ground. See http://isp-lists.isp-planet.com/isp-clec/ for details and archives. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avaya Partner Phones to SIP?
Title: Avaya Partner Phones to SIP? I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific experiences. Would anyone care to enlighten me? Off list is fine if this is a repeat, thanks very much. Matt
[Asterisk-Users] Re: Grandstream tftp cfg.txt format
Maron Kristófersson wrote: Hello! I've been reading through the archives on this list for the last 8-10 months. There are some reports on success with tftp autoconfiguration with a given cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that creates such a file, but I need the format to do that. Regards, Maron Kristofersson I suspect that the txt version of the cfg file is used as input to GAPS, not input to the phone. What we really need is a handful of working cfg files that have already been compiled by GAPS into the loadable binary format that the phone probably wants. If each file had only one item changed, and were accompanied by a detailed description of the corresponding phone setup, then the file format could be decoded, of course, providing that the file format is not encrypted (I doubt this, however). The problem will be getting the required files. GS asked me to sign an NDA before they would even consider letting me experiment with their GAPS system. I suspect that everyone else has had to do the same. Stephen R. Besch ___ 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] G729 codec for asterisk
On Thu, May 20, 2004 at 11:17:55AM -0500, brian said: I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. No you MUST pay per channel because the patent holders require that. The patent holders would [EMAIL PROTECTED] kittens if you had no port limit or any type of control on it. That's why the control and registration processes are in place to comply with the patent holders requirements. So your request translates into I want something for nothing. I read it more like he want's to purchase a site license, not a totally unusual request. If used within a restriced environment (no resale) it's reasonable, although would be expensive. Probably something he would need to negotiate with the patent holders directly at this point. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 200 and hold
First, try moving back to 2.05c or earlier. 2.05e has a few problems (remember, it's beta quality) that could be causing this. Second, are you sure that the disconnect on hook or transfer on hook settings are the way you expect them to be. That caught us for a while since we were putting people on hold and then putting the phone on hook, which had the result of disconnecting them. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Thursday, May 20, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom 200 and hold Hi, I've looked through the archives and seen references to placing calls on hold on a snom 200 (any version of the firmware but we have the latest: 2.05e.) Basically, we can't place calls on hold on the snom 200! The manual talks about the Flash button (which is really the R button, as far as I can tell.) Pressing the R button will immediately disconnect the incoming call. Another poster to this list indicated one could just choose another line and the current line will be put on hold. This is not true on our phone: again, the original call is immediately disconnected. We've been all over the settings in the snom 200 and have tweaked a bunch of parameters. So: how does one place an incoming call on hold on a snom 200 so that we can do attended transfer? Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] VoicePulse broken?
Inbound is working here, no problems that I know of. Scott - Original Message - From: C. Sullivan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 12:52 PM Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ 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] Anonymous sip register
Chad Brown wrote: Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? Im thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Just configure the context= in the general section of sip.conf to point to a context in extensions.conf that accepts calls and directs all of them to ivr. /O ___ 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] G729 codec for asterisk
The Wiki is a bit wrong.. you can record raw g729 streams to disk, what do you think format_g729.c is? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Thursday, May 20, 2004 11:25 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G729 codec for asterisk This is discussed at length in the Wiki, on several pages, including: http://www.voip-info.org/wiki-Asterisk+G.729+licensing Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of pesb Sent: Thursday, May 20, 2004 8:59 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 codec for asterisk Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could we make some sort of agreement on having a no-limit channel license? Even, we would like to have the possibility of installing it on how many machines we wish to do. I am really in a hurry here. Please answer as soon as possible. Pablo Salinas RD - CONEXION S.A. Asuncion - Paraguay (S. America) Phone:595-21-440104 Fax: 595-21-440270 ___ 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] VoicePulse broken?
HAHAH why do they ever work! Take this to the -biz list please! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Sullivan Sent: Thursday, May 20, 2004 11:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ 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] G729A problem
Hi all, Unable to find translation path. How to fix ? May 20 18:22:49 NOTICE[1224059824]: channel.c:1508 ast_set_read_format: Unable to find a path from G729A to ULAW May 20 18:22:49 NOTICE[1224059824]: channel.c:1478 ast_set_write_format: Unable to find a path from ULAW to G729A May 20 18:20:47 NOTICE[1232452528]: channel.c:1508 ast_set_read_format: Unable to find a path from G729A to ULAW May 20 18:20:47 NOTICE[1232452528]: channel.c:1478 ast_set_write_format: Unable to find a path from SLINR to G729A
Re: [Asterisk-Users] Avaya Partner Phones to SIP?
Matthew Branton wrote: I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific experiences. Would anyone care to enlighten me? Off list is fine if this is a repeat, thanks very much. Matt http://www.citel.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
