Re: [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?
Have you tried the altarch variant of centos for 32 bit? You could also potentially use centos 6, though that is going eol soon as well .. Kris Stark Sent from from a mobile device. From: Ira Sent: Sunday, June 24, 2018 03:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora? Hi, I’m currently using 32 CENTOS 5 but it’s now unsupported. I only have a 32 bit processor and CENTOS no longer supports 32 bit so I need to move on. I’ve installed the current version of 32 bit Fedora and I can’t get the latest Dahdi to build. Even tried downloading the early release DAHDI from github with no luck. Any recommendation for which 32 bit LINUX to use going forward? Or optionally, how to compile Dahdi on the most current Fedora. The error was something to do with xxx.timer.timer.xxx = xxxtimer_xxx I don’t have the error at hand, because I had to put the old drive back in to get our phones working again. Thanks, Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold
OK - so somebody just handed me the new music on hold file to use for the organization... Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. In general, we don't ever see a user being placed on hold for more than a minute, so using this file directly is of no use in general if I were to place it directly in to the server, as all users will only hear the first little bit of it. I suspect that when this was created, the producer assumed that the file would play in a loop, starting and stopping as callers were on hold. I realize that the streaming category will do just that, but since this is a local file, the setup works differently. (This is replacing a set of about 10 previous files that worked perfectly.) Is there any way, other than splitting up the file and trying to make decent segues between the files, to get this to work on a current version? I realize that getting it redone would be the best way, but I don't know if that is going to be an easy possibility. Any recommendations? Thanks! Kris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys PAP2-NA failures...
Has anybody else experienced problems with the Linksys PAP2-NA's? I've now had two of them fail unexpectedly, with no apparent rhyme or reason, having gone into a RED power LED, with a solid blue ethernet LED. No response from the device either on the network or from the phone To make matters even crazier, the one that now failed was the one I received as a replacement for the previous dead one - and no, they were never installed in the same location either Grrr Kris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Devenijn wrote: | Fine but don't mix up Swedish Danish beer ... How could you - swedish beer? I barely think (with my apologies to the swedes) that it is even beer for the most part... The regular joke is that it is light oil from the l and how well it can sound like the finnish word for oil *grin* Kris | | -Oorspronkelijk bericht- | Van: [EMAIL PROTECTED] namens steve szmidt | Verzonden: vr 1/04/2005 16:39 | Aan: Asterisk Users Mailing List - Non-Commercial Discussion | CC: | Onderwerp: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now | | | | On Friday 01 April 2005 02:40, Olle E. Johansson wrote: |During the developer's conference call yesterday evening, |it was decided that we finally should release the much-awaited |Asterisk 2.0 Stable release, also called codename AAFJ. | | Olle, you better take a break! | | For the rest of you, good luck! You'll need it. I think finally the Danish | Elephant beer that is so strong has gone to Olle's head. | -- | | Steve Szmidt | | They that would give up essential liberty for temporary safety | deserve neither liberty nor safety. | Benjamin Franklin | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCTfZDga76cWglRW4RAswhAJ4+9JwTNWp/MJy/Zzi+nQA1nc8LIwCffPBu Qufewc/4Gg64sldsOG4hkNM= =bcGB -END PGP SIGNATURE- begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toll Free dialing problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Why not juse use an ENUM lookup for those calls? There is info on the list archives (http://lists.digium.com/pipermail/asterisk-users/2004-January/034423.html) about using that, and I know it was elsewhere too... Works just fine... Kris Shadow Roldan wrote: | Ok so I found these threads in which some people have had similar issues | but no solutions yet... | http://lists.digium.com/pipermail/asterisk-users/2003-April/009595.html | http://lists.digium.com/pipermail/asterisk-users/2005-March/095725.html | | | I've tried using iaxtel and BroadVoice to route toll free calls and the | call appears to connect ok (see log snippet below) but it just rings and | rings and eventually it times out and I get | | The person you are calling is unavailable | | | The really dumb thing is all the numbers I am trying always pick up on | the first ring or without a ring when dialing from the PSTN. | | | Any ideas? | | | Log snippet below: | | -- Executing Dial(SIP/116-3e81, | SIP/[EMAIL PROTECTED]|45) in new stack | -- Called [EMAIL PROTECTED] | -- SIP/sip.broadvoice.com-ace8 is making progress passing it to | SIP/116-3e81 | -- Nobody picked up in 45000 ms | -- Executing Congestion(SIP/116-3e81, ) in new stack | == Spawn extension (inside, 18887467426, 2) exited non-zero on | 'SIP/116-3e81' | | | _ | | Shadow Roldan | IT Manager | Zero G Software, Inc. | mailto:[EMAIL PROTECTED] | www.ZeroG.com | | The leading provider of multiplatform software deployment solutions. | _ | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCS1Shga76cWglRW4RAsujAJ9ctZnBbRNWAQdyYWhni6g7brJUfgCfezgv 3GSbAdgICTtStXUrK3Dcv0s= =jWa6 -END PGP SIGNATURE- begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom phones-buggy SIP firmware or am I missing something in the XML configs?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jason Brown wrote: | Anyone have experiece with polycom phones? | | I am experiencing a really weird problem. In an office where I have the | following extensions: | On the Polycom phones, when I want to dial from extension 100 to any | extension 120 or above, or dial out, it dials just fine. If I want to | dial from extension 100 to extension 101,or 102 or 103 or 104, after you | dial 10 then it flashes connecting (really fast flash) but doesn't | connect to anything. Then you can dial the last digit of the extension. | Otherwise, if you dial 101 you are forced to dial the last 1 twice | because it wont send it. | | I have ruled out asterisk completely. Nothing wrong in the dialplan. I | have also ruled out DTMF. So it can either be buggy firmware or | something I am missing in the XML configs. Phone dialplan rules seem to be the culprit for something like that. Take a look at what the phone has set as the rules, and set as appropriate. Instructions are available, a link was posted earlier on the list... Kris | | Any ideas? | | | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCQj87ga76cWglRW4RAmJCAJ9K1sWopwoiUuKa9FfmUbPh3A/hNACaAwtu o3B+MKIkRGhxH8lgnLfMB6Y= =ih6e -END PGP SIGNATURE- begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit the call recording when pressing *1
Joseph wrote:
On Sat, 2005-02-26 at 22:14 -0800, Luki wrote:
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
I think should be:
exten = 21,1,Dial(${phone1},20,r,L(5:4:1))
The [] mean the parameter is optional, but you don't use them when
specifying the values.
I've tried that too, doesn't work!
5min. has passed and the call wasn't disconnected nor I hear any warning
to message how many minutes are left.
Options should be in a single section so that all options are
delimited by a single comma. Thus, try:
exten = 21,1,Dial(${PHONE1},20,rwL(5:4:1))
Kris
begin:vcard
fn:Kris Stark
n:Stark;Kris
org:Dataflow
adr:Suite B;;401 E State St;Ithaca;NY;14850;USA
email;internet:[EMAIL PROTECTED]
title:IT Manager
tel;work:+1 607 272 8589
tel;fax:+1 607 272 8634
tel;cell:+1 607 768 4401
x-mozilla-html:FALSE
url:http://www.goDataflow.com
version:2.1
end:vcard
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Re: [Asterisk-Users] softphone has problem to call out via X100P card
Ho Chan wrote:
Hi all,
I have the Asterisk set up and 2 softphone (Xlite) set up on two other
PC. With the following configuration, I can use one softphone (2000) to
call the other one (2001) and/or the voicemail at 2999.
Here is my problem:
1. When I dial 9+nxxx- with one of the softphone to the PSTN
via X100P card, I got busy tone. (i.e. I want to use the phone line
which is connected to the X100P to call out)
You need to allow the softphones to go to the context that allows
dialling out. See below - inline comments...
2. When I use my cell phone to call the phone line which is
connected to X100P, it just rings for 4 times then hang up on me. (i.e.
Asterisk never answer the phone)
You need to define a context for the zap inbound channel to go to - see
below...
Zapata.conf
language=en
[snip]
channel=1
context=from-sip
The above is not quite recommended, but would work with the config that
you have... Ideally, you'd have an incoming context in your
extensions.conf, and the context in zaptel would then point into that
context.
-
Sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
allow=all
context = outgoing
[2000]
type=friend
[snip]
context=from-sip
mailbox=100
This sends all your calls from the softphone into the from-sip context
in extensions.conf.
Extension.conf
[general]
static=yes
writeprotect=yes
[outgoing]
ignorepat = 9
exten = _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t)
exten = _9NX.,2,Congestion
None of your calls ever get into this context... You would want to have
a include = outgoing in your from-sip context to be able to dial out.
However, with the way your extensions.conf is set up right now that
would be a security risk since to be able to call in you would have to
have your inbound calls from the PSTN going into this same context...
[from-sip]
exten = 2000,1,NoOp(call for ${EXTEN})
exten = 2000,2,Dial(SIP/2000,20,tr)
exten = 2000,3,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
[snip]
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
; Call straight to extension 2001
exten = s,1,Answer
exten = s,2,Dial(SIP/2001,20,tr)
exten = s,3,Voicemail(u2001)
exten = s,4,Voicemail(b2001)
Basically, try something like:
[inbound]
; Since you want all your calls from the PSTN to go to ext 2001
exten = s,1,Answer
exten = s,2,Dial(SIP/2001,20)
exten = s,3,Voicemail(u2001)
exten = s,103,Voicemail(b2001)
[outbound]
ignorepat = 9
exten = _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t)
exten = _9NX.,2,Congestion
[from-sip]
include = outbound
exten = 2000,1,NoOp(call for ${EXTEN})
exten = 2000,2,Dial(SIP/2000,20,tr)
exten = 2000,3,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2001,1,NoOp(call for ${EXTEN})
exten = 2001,2,Dial(SIP/2000,20,tr)
exten = 2001,3,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
Kris
begin:vcard
fn:Kris Stark
n:Stark;Kris
org:Dataflow
adr:Suite B;;401 E State St;Ithaca;NY;14850;USA
email;internet:[EMAIL PROTECTED]
title:IT Manager
tel;work:+1 607 272 8589
tel;fax:+1 607 272 8634
tel;cell:+1 607 768 4401
x-mozilla-html:FALSE
url:http://www.goDataflow.com
version:2.1
end:vcard
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Re: [Asterisk-Users] Updating Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Blair wrote: | | I initially installed Asterisk from a vendor supplied CD. I want to | maintain a more current release so I am trying to update from CVS. | I removed the previous vendor's release and followed the | instructions you provided. I got the following error. Can you | explain why? | | Thanks, | Steve | | --- cut here --- | | isk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ | -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN | -c -o channel.o channel.c | channel.c:49:2: #error You need newer zaptel! Please cvs update zaptel ~ ^^ Right there - update zaptel, recompile and install, then recompile (and/or update again) asterisk itself. Keep in mind that you have to have all the dependencies updated in sync with asterisk... K | channel.c: In function `ast_channel_alloc': | channel.c:303: `ZT_TIMERPONG' undeclared (first use in this function) | channel.c:303: (Each undeclared identifier is reported only once | channel.c:303: for each function it appears in.) | channel.c: In function `ast_queue_frame': | channel.c:418: `ZT_TIMERPING' undeclared (first use in this function) | channel.c: In function `ast_read': | channel.c:1244: `ZT_EVENT_TIMER_EXPIRED' undeclared (first use in this | function) | channel.c:1246: `ZT_EVENT_TIMER_PING' undeclared (first use in this | function) | channel.c:1255: `ZT_TIMERPONG' undeclared (first use in this function) | make: *** [channel.o] Error 1 | --- end cut -- | | [EMAIL PROTECTED] wrote: | | Just tried it. Show version still shows: | | Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 | | | | Well, only thing I can see is that your CVS download didn't went | right, or you downloaded it into a different place, because you're not | even at 1.0.4 | | Follow these simple steps to update you tree : | | # cd /usr/src | # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot | # cvs login- the password is anoncvs. | | # cvs checkout -r v1-0-5 asterisk | # cd asterisk | # make clean; make | | then, stop asterisk | | # make install | | then start asterisk | | HTH | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFB9oTVga76cWglRW4RAsboAJ9XSymy4i1Ldx6AxuFCDbM9l1rsIACgnMJ7 R9hrvj3SHfp9XJdlqpX6hLo= =Mw0N -END PGP SIGNATURE- begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272-8589 tel;fax:+1 607 272-8634 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fwd IAX2 error
Joseph wrote: I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (40) of length 2 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:3044 authenticate_verify: requested inkey 'freeworlddialup' for RSA authenticationdoes not exist You defined the authentication to use the public key, freeworlddialup.pub, and asterisk cannot find it. From the FWD web page: You will also need the freeworlddialup.pub key in your /var/lib/asterisk/keys/ directory. (If you installed from cvs after June 3, you probably have this file already.) If not, you can wget it from this link. The link is http://downloads.fwdnet.net/freeworlddialup.pub Kris When I try to call out I get: Called 495771:[EMAIL PROTECTED] WARNING[114696]: chan_iax2.c:4534 socket_read: Call rejected by 65.39.205.121: No such context/extension begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tall free number via FWD over IXA2
Joseph wrote: I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called x:[EMAIL PROTECTED]/18007425877l -- Call accepted by 65.39.205.121 (format ULAW) -- Format for call is ULAW -- Hungup 'IAX2[65.39.205.121:4569]/3' Dialing *1800... doesn't do anything. What am I doing wrong? The number that is passed to FWD needs to be in the format *1800..., so you would want to make sure your dialplan is set up such that it passes the number with an * at the beginning. Alternatively, use one of the ENUM lookups to do toll free instead - look up ENUM on the wiki. Kris begin:vcard fn:Kris Stark n:Stark;Kris org:Dataflow adr:Suite B;;401 E State St;Ithaca;NY;14850;USA email;internet:[EMAIL PROTECTED] title:IT Manager tel;work:+1 607 272 8589 tel;fax:+1 607 272 8634 tel;cell:+1 607 768 4401 x-mozilla-html:FALSE url:http://www.goDataflow.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Polycom IP 400
Sorry for the slightly OT question... Anybody have configuration files for a Polycom IP 400 MGCP version, or be able to point me in the right direction for being able to create them? Thanks! Kris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: G729 Segmentation fault
On Tue, 2004-05-18 at 22:55, Jeremy Bogan wrote: You must register the codec in order to be able to use it. May 12 19:27:42 WARNING[16384]: codec_g729b.c:511 load_module: Unable to initialize va stuff: -1 Segmentation fault alberspilnx8:/bin # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe I'm getting the same errors. I've registered my codec license and also installed the certificate from voiceage, but no dice. Has anyone else experienced this problem? Are you using the safe_asterisk script to start up? G729 requires a tty, which the script provides - at least so I've read... I can get mine to segfault every time if I start up using just the asterisk command, safe_asterisk works every time... Kris Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dialing 800 numbers with VOIP
Matt Lawson wrote on Mon, 09 February 2004 16:55] Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? No difficulty on this end to dial 800 numbers via Voicepulse... I just don't do so normally - as others stated, I either use IAXTEL or FWD for that... On a different note - is something up with the freenum.org enum lookups? I've had them fail on me lately - or did I miss the boat on some announcement? More accurately, I've had them fail on all US numbers... Kris ___ 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] Introducing Firefly
Wow! That seems very reasonable for a phone that looks like it will *work* I wish I could say I was a reseller to get an early sample, but I guess I'll just have to wait with baited breath like most of us here... :) Kris On Fri, 2004-01-30 at 00:05, Adam Hart wrote: We looking at a street price of around $99. We are aiming for 6 weeks to send various resellers samples. (email [EMAIL PROTECTED] to obtain a sample) It shouldn't be too longer after that till we have our first production run. - Original Message - From: Chris Albertson [EMAIL PROTECTED] When will the IP phone be available and would you have an idea about the price? --- Samuel Jimenez [EMAIL PROTECTED] wrote: Nice!! Have just tried it a bit, seems cool... Congrats!!! Will test it against my * box and will provide some feedback. Thanks! Sam\\\ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kris Stark -- IT Manager Dataflow / Russell Rhoades Reprographics / National Direct Reprogrphics 401 E State St, Suite B Ithaca, NY 14850 (607) 272-8589 ___ 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] Prefix the * character
Any ideas on how to do this one? FWD requires an * on certain calls as a prefix character, but I cannot seem to be able to get Prefix(*) to add that to the front of the extension that is dialed... Setting up an extension that dials (SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the prefix to a dialed number, it acts as if the Prefix command was not even there, ie: exten = 1,1,Prefix(*) exten = *1,2,Dial(SIP/[EMAIL PROTECTED]) would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED] Any ideas? Tnx Kris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefix the * character
On Sun, 2003-12-07 at 23:57, Tilghman Lesher wrote:
On Sunday 07 December 2003 22:48, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I
cannot seem to be able to get Prefix(*) to add that to the front of
the extension that is dialed... Setting up an extension that dials
(SIP/[EMAIL PROTECTED]) works just fine, but in trying to add the
prefix to a dialed number, it acts as if the Prefix command was not
even there, ie:
exten = 1,1,Prefix(*)
exten = *1,2,Dial(SIP/[EMAIL PROTECTED])
would dial [EMAIL PROTECTED], and not [EMAIL PROTECTED]
Any ideas?
Prefix is an older application which was more useful prior to being
able to manipulate variables (the days of BYEXTENSION instead of
${EXTEN}). Instead, do:
Dial(SIP/[EMAIL PROTECTED])
-Tilghman
Thanks! I actually had tried this to no avail, but then noticed after
digging through that it was in fact using the wrong entry to begin
with... An simpler match was being made, and so it never got to that
part of the dialplan...
Thanks again!
Kris
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[Asterisk-Users] Sipura / Handytone / Cisco
Could anybody shed some light in which device they would use in this situation: Remote office PBX's to be connected via a) Cisco ATA-186 or b) Sipura SPA-2000 or c) Grandstream HT-ATA-286 to go via the net to an * box. Pros / Cons for each device would be appreciated! Thanks Kris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
