RE: [Asterisk-Users] Looking for help with a Polycom Soundpoint IP 600
This can happen if the (mac-addr).cfg file is bad. The first time you load the phone it will corrupt the flash and you will have to send the phone in for repair. Make sure when you edit the file you don't add any carriage returns... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryeh Sent: Sunday, January 16, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Looking for help with a Polycom Soundpoint IP 600 Hi all, I have a Polycom Soundpoint IP 600 that looks like it is fried. It either has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt that asks if I want to enter the setup configuration or continue booting. When plugged into power, it turns on, shows the Polycom logo for 3 seconds and then pretty much goes dead with two red lights on. The one in the top right corner and the one in the top left (the line 1 red light). Anyone have any clues as to what to do? I'm looking into finding a polycom service center or reseller that is in or close to New York City. Anyone know of any places that can simply connect up to the phone and reflash the rom and firmware. I'm guessing it's a 5 min fix with the right equipment. Any help would be appreciated. Thanks guys. aryeh ___ 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
[Asterisk-Users] quadBRI asterisk error message message: "not able to open Zap channel"
Hello,
I cannot make any call with my new quadBRI card
from Junghanns.net in my asterisk box.
My asterisk box is built on Fedora Core 3 linux
system. After compiling drivers from junghanns.net the card driver was loaded
correctly with
- modprobe zaptel
- insmod qozap.ko ( as FC3 is running
a 2.6 linux kernel)
- ztcfg -vvv ( shows 12 channels
correctly configured and the 4 leds on the card were green : ISDN layer 1 was
working)
- the 4 spans are configured in TE
mode
- my ISDN provider is France Telecom
- i didn't make major changes in
the zaptel.conf and zapata.conf files provided with the bri-stuff package
downloaded from Junghanns.net :
zaptel.conf :
loadzone=fr
defaultzone=fr
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
zapata.conf
[channels]
;Default
language
;language=en
switchtype =
euroisdn
; p2mp TE
mode (for connecting ISDN lines in point-to-multipoint
mode)
signalling =
bri_cpe_ptmp
pridialplan
= local
prilocaldialplan = local
echocancel =
yes
context=from-pstn
group =
1
; S/T port
1
channel
=> 1-2
group =
2
; S/T port
2
channel
=> 4-5
group =
3
; S/T port
3
channel
=> 7-8
group =
4
; S/T port
4
channel
=> 10-11
extensions.conf
exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
But when trying to give a call, i'm always
receiving " not able to open Zap channel" from my asterisk box ..
Am i using the right parameters for france telecom
?
Any help would be apreciated !
Thanks.
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[Asterisk-Users] voicemail attach not in 1.0.2 ?
Hi In voicemail.conf I have attach=yes (tried with =1 and = thrue) but I cant get asterisk to attach the voicemail. Any clue ??? (using ast_data) /HHA ___ 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] IAX1 vs. IAX2
Joseph wrote: Joseph, 1 - 0.9 still uses IAX2 (I think - pretty sure). 2 - Why are you using 0.9? Maybe they should be the other way around... I'm just using default installation whatever Gentoo is providing; this is their stable version. Joseph, While I also use Gentoo(as do many others), most will tell you NOT to install * from portage. You can save yourself trouble by getting 1.0.3 or CVS and ditch the builds from portage. -- Kristian Kielhofner ___ 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] Meetme conf and Shoutcast
On Sun, 16 Jan 2005, Mike wrote: We would like to know if there is a way to broadcast (in realtime) a conferance. http://www.voip-info.org/wiki-Asterisk+cmd+Ices I haven't tried it though. -- Regards, Tobias Jönsson, Lund SE___ 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] Re: SS7 and Asterisk solution
On Sun, 16 Jan 2005, Dorn Hetzel wrote: > On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote: > > > > You can modify and/or link to GPLed code with commercial code and get > > away with it as long as you don't distribute the stuff. That's the > > story with G.729, with nVidia drivers etc etc etc The g.729 license for Asterisk is special - there is an exception in the asterisk license I think. The nVidia driver is considered to not be a derived work by the powers that be (Linus et al) since they specifically allow binary modules using only the published api. > I suppose it's even possible to distribute your commercial code in source > form and ask your customer to acquire their own copy of * to link it with. > (is that actually true?) Probably not, but you have to ask a lawyer. Your code may or may not form a derived work in the legal sense. An analogy is that you can not distribute an alternate ending to a book by writing the last two chapters and distributing them. That would form a derived work. For software the lines are blurry and untested. Peter ___ 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] Any interest in a Canadian Asterisk mailing list?
Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or [EMAIL PROTECTED] Cheers! John ___ 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] pattern matching problem
How do I solve the problem with between patterns:
_1800
_1NXX
I would like all numbers 1800, 1877 etc to go through iaxtel
but all other numbers 1xxx via voipjet
Example in my extension.conf I have:
[iaxtel]
exten => _1700NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _1888NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _1877NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _1866NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _1800NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
[outgoing-voipjet]
exten => _1NXXNXX,1,SetCallerID(4757894789);
exten => _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Whatever I try to dial it goes through voipjet.
--
#Joseph
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[Asterisk-Users] Asterisk over External Motorola BitSurfR Pro ISDN Modem
Hi, I have an external Motorola BitSufR Pro ISDN modem and an ISDN BRI line. Is that possible to get this to work with Asterisk for dial in/out? Somebody ever did this? Where should I start? Thanks in advance, Stephane ___ 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] Which is better IP500/IP600 or /CP7960
polycom is better for the same quality and lower price. On Sun, 16 Jan 2005 17:27:20 -0800 (PST), Robert Augustyn <[EMAIL PROTECTED]> wrote: > Any preferences? > And why? > Thanks in advance. > robert > ___ > 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
[Asterisk-Users] Registering with IAX provider
I have in iax.conf register => name:[EMAIL PROTECTED] but I can not make a call, it hangs up on me. How can I check if I'm registered with iaxtel? What do I have to have in iax.conf in order to register? -- #Joseph ___ 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] IAX1 vs. IAX2
> Joseph, > > > 1 - 0.9 still uses IAX2 (I think - pretty sure). > 2 - Why are you using 0.9? > > Maybe they should be the other way around... I'm just using default installation whatever Gentoo is providing; this is their stable version. -- #Joseph ___ 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<->NAT<->*
> Making asterisk work through NAT is a pain and some of the Wiki stuff> is wrong/out dated. This works for me:Please feel free to fix or point out what is wrong/outdated so someone else can fix. Thanks. ___ 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] Looking for a VoIP provider for my Asterisk box. {Scanned}
Hello All, I have Vonage and Lingo and like the service, but I would like to drop there ATA equipment. I tried BroadVoice had them for less then 24hrs. Anyways I would like to connect Asterisk directly to a VoIP provider without the use of there ATA equipment. Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact [EMAIL PROTECTED] if you have questions about this email. ___ 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<->NAT<->*
You probably want to use IAX to talk to FWD. It tunnels through NAT without any special changes. See http://www.fwd.pulver.com/advanced/iax Making asterisk work through NAT is a pain and some of the Wiki stuff is wrong/out dated. This works for me: In sip.conf: localnet: 192.168.1.0/255.255.255.0 externip: Then look at rtp.conf and see what range of ports it uses. Mine is 1-11000 (since I never have a lot of calls at once) Then I configured my firewall to route 5060 (sip) and 1-11000 (rtp) to my asterisk box. I'm sending both TCP and UDP, but I suspect UDP is really the only protocol required. - |Daryll ___ 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] IAX1 vs. IAX2
Joseph wrote: I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf is set port=5036 Can I register with a provider who is using IAX2 ? When I set it up and run: iax2 show registry - it is not displaying any registered provider. Joseph, 1 - 0.9 still uses IAX2 (I think - pretty sure). 2 - Why are you using 0.9? Maybe they should be the other way around... -- Kristian Kielhofner ___ 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] IAX.conf error
Joseph wrote: When loading iax.conf I get warning: WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now In iax.conf I have: [general] port=5036 Joseph, You are probably going to want to change that to 4569 anyways... -- Kristian Kielhofner ___ 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] New Sipura-841 phone.Mike volume problem.
Sounds like you've got a problem with your microphone. I got my SPA-841 a while ago and the microphone works just fine. I don't have to scream into it. I like the phone a lot. I agree with you that the buttons have a somewhat odd feel. They're sort of rubbery and don't slide like plastic ones, but they do work just fine. Another thing that bothered me was that there was no documentation with the phone. I mailed support about it and they said the docs are due out early next week and would be available on the support website. The setup is very much like the other Sipura devices, so the existing ATA documentation covers most of what you need. I'm curious about things like custom ring tones, and displaying things on the LCD which I hope the real manual will explain. The other question I asked was about intercom support. They claim it's on their feature list and should arrive in a firmware update some time this quarter. That would be a really useful feature for the phone. The MWI light works with Asterisk. There's a voicemail button you can program to call whatever number gets you to your voicemail. So those work nicely together. Overall it's got a nice set of features for a business phone at a very reasonable price. - |Daryll ___ 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] TDM04B vs Dell
On Wed, 2005-01-05 at 16:01 -0800, Michael Swan wrote: > Hi all, > > I've struggled for several days trying to get a Digium TDM04B 4-port > wxfco card working on a Dell 1U PowerEdge 750 machine running > Fedora Core 1. I finally got a call back from Digium who indicated that > there is a fundamental conflict between the card and the PowerEdge > having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. Sorry to revive this older thread, but I thought it might be of interest to add a new data point into this. I have a Dell 750 with a T100P card in it now and it is seeing interupts just fine. Granted it is missing some from time to time on a system not even running asterisk yet. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] New Sipura-841 phone.Mike volume problem.
Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press them they feel like they are dirty with some sticky stuff on them. They don't get stuck but feel that way. Here is my problem. The mike is really bad on the phone. It's not the hand set or the plugging via the 2.5 plug on the side. It's something to do with the phone hardware internal. I can tap on the mike and I hear a faint tap on the other end. But unless you scream into the handset or mike they can't hear you. I need to see if there is some type of fix for this. Registration and setup is just like all of the Sipura devices via the web. In fact most of the setting are almost line by line like the Sipura 2100. Looks great on how they did that. ___ 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] FWD<->NAT<->*
I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register => FWD#:[EMAIL PROTECTED]/FWD# bindaddr=0.0.0.0 - is this the address of my asterisk server externip=xxx.xxx.xxx.xxx - this is my external IP before firewall isn't it? When I try to register, I get: sip_reg_timeout: Registration for ... What am I missing? -- #Joseph ___ 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] Meetme conf and Shoutcast
We would like to know if there is a way to broadcast (in realtime) a conferance. We hold large phone conferances and would like to know if we could have some of our users listen over a streaming services. Formats we have looked at include: Shoutcast,Real Networks,QuickTime, and dare I say Windows Media player. The issue we have, is that I can not find a way to transfer the stream in realtime to one of these formats. We would like to use shoutcast, has anyone have any idea how to do this? Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP - INBOUND Call - best setup
What would be my best option to receive calls via VOIP. I would like to use it as an alternative number when my main number is busy. The solution is not that easy as in order for customer to be a free call DID=Direct Inward Dialing provider would need to be a local company, I think. Correct my anybody if I'm wrong. I'm located in Alberta Canada so my chases are even smaller. I've another incoming fax line, so I guess I could set it somehow as an alternative incoming line if my main line is busy. -- #Joseph ___ 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] Which is better IP500/IP600 or /CP7960
Any preferences? And why? Thanks in advance. robert ___ 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] No compatible codecs
I've heard problems with the Grandstream G729 and the new digium G729 by MAC ID. Could be a compatibility issue with the implementations. Did you ever use the Grandstream against asterisk with the old Voiceage G729? I've heard that works just fine. -- William ___ 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] IAX.conf error
Joshua Colp wrote: This is person normally and it is NOT AN ERROR. :) Dats grate england you have they're... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] IAX2 one side loses audio
I am afraid i do not have a solution for you, but we also had this problem occur, exactly the same. It happened overnight, with no changes to the server. With help from our IAX provider, we did many tests, no solution, we then moved to a SIP connection to our provider, problem solved. Our * server is a beast with 2GB DDR ram and no load, QOS, 2MB leased line.. Ask your provider if you can try SIP with them. From: [EMAIL PROTECTED] on behalf of Trevor Peirce Sent: Sun 16/01/2005 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX2 one side loses audio It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists. Thanks, Trevor Peirce ___ 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
[Asterisk-Users] X100P with no sound!
Hi, can't get
X100P (fully zapata compatible clone) to work (I'm in Australia).*
recognises the card and the channel (1) but has definetely some problems
talking to the pots line.I set up this simple dialplan for ZAP
("incoming" context, as setup in zapata.conf, for channel
1)[incoming]exten => s,1,Answerexten =>
s,2,Playback(somefile)exten => s,3,Goto(default|266|4) ; to get DISA
accessexten => _13.,1,Dial(Zap/1/${EXTEN}) ; to call 1300 numbers in
AustraliaWhen I ring my home number, connected to the x100p Asterisk
picks up and performs all priorities, but I hear nothing. CLI show all
applications being performed (answer, playback of file, call of
extension...).These are my
setting:indications.conf[general]country=au[au]description =
Australiaringcadance = 400,200,400,2000dial = 425*25busy = [EMAIL PROTECTED],[EMAIL PROTECTED];10(.375/.375/1+2)ring = 425*25/400,0/200,425*25/400,0/2000; XXX Congestion:
Should reduce by 10 db every other cadence XXXcongestion =
400/375,0/375callwaiting = 425/100,0/100,525/100,0/4700dialrecall =
!425*25/100!0/100,!425*25/100,!0/100,!425*25/100,!0/100,425*25record =
1400/425,0/14525info = 400/2500,0/500zapata.conf
:[channels]language=encontext=defaultswitchtype=nationalusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yescancallforward=yesmailbox=relaxdtmf=norxgain=10.5txgain=-4.5group=1callgroup=1pickupgroup=1immediate=nobusydetect=yesbusycount=6callprogress=yessignalling=fxs_ksechocancel=yesechocancelwhenbridged=yesechotraining=400callerid=asreceivedgroup=1context=defaultchannel
=> 1I'm running * on a Pentium II, 350Mhz, with 198 RAM (soon will
add 128).* also has a prob detecting hangup. Each time I test the the
zap channelthe phone line remains busyThanks for the
help,hope someone knows what's
happening!Manny
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RE: [Asterisk-Users] IAX.conf error
This is person normally and it is NOT AN ERROR. It just states that it's ignoring the port. Simple as that? Okay? Okay? Everyone repeat after me: WARNINGS ARE NOT ERRORS. Thank you. - Joshua Colp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Sunday, January 16, 2005 7:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX.conf error When loading iax.conf I get warning: WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now In iax.conf I have: [general] port=5036 -- #Joseph ___ 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
[Asterisk-Users] IAX.conf error
When loading iax.conf I get warning: WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now In iax.conf I have: [general] port=5036 -- #Joseph ___ 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] Re: SS7 and Asterisk solution
On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote: > > You can modify and/or link to GPLed code with commercial code and get > away with it as long as you don't distribute the stuff. That's the > story with G.729, with nVidia drivers etc etc etc > I suppose it's even possible to distribute your commercial code in source form and ask your customer to acquire their own copy of * to link it with. (is that actually true?) -dorn ___ 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] H323 Softphone for iPAQ
http://www.sjlabs.com/sjp.html SJphone® is a VOIP softphone that allows you to speak with any PC, PDA, stand-alone IP-phone and with any legacy wired or mobile phone (using your VOIP gateway or purchasing service from Internet Telephony Service Provider). It supports both SIP and H.323 standards and is fully interoperable with most major IP-telephony vendors and ITSP. I’m just about to try it my self ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Sunday, January 16, 2005 8:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] H323 Softphone for iPAQ Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid ___ 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] IAX1 vs. IAX2
I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf is set port=5036 Can I register with a provider who is using IAX2 ? When I set it up and run: iax2 show registry - it is not displaying any registered provider. -- #Joseph ___ 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] No compatible codecs
Andres, Thanks for your answer, but as you can see in the output from show translation in my original post my Asterisk DOES have G729 support. Also the fact that softphones work but the Grandstream does not work stumbles me. Rene Kluwen Chimit - Original Message - From: "Andres" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 16, 2005 11:46 PM Subject: Re: [Asterisk-Users] No compatible codecs > > >Any suggestions about what I can change to make this work? > > > > > Yes, you should get a G729 license for your Asterisk. > > >Cheers! > > > > > > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No compatible codecs
Any suggestions about what I can change to make this work? Yes, you should get a G729 license for your Asterisk. Cheers! ___ 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] Inbound Callerid for SIP Phones
Hi Michael On 16 Jan 2005, at 20:22, Michael Johnston wrote: Currenly the inbound lines do not have callerid on them so callerid=no in my zapata.conf file. What happens on inbound calls is that the SIP extensions are dialed but their callerid shows '[EMAIL PROTECTED]:X.com'. Does anyone know how to change the callerid on the inbound calls? You can do this using this kind of setup in extensions.conf: (Put this in your inbound context which you defined in zapata.conf.) exten => s,1,SetVar(ALERT_INFO=Bellcore-r2) exten => s,2,SetCallerId,Line 1 <1> exten => s,3,Dial(SIP/110&SIP/111,20,tr) Phil. -- Phil Quinney IT Consultant - 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
RE: [Asterisk-Users] TE410P problem (Looping UP Span 1...) [digium.com #13999]
The the 'common' factor here appers to be the Intel E7520 Chipset. I have a NEC 120Rg-2 here with this chipset with the same problem. This chipset exists in the HP DL380 G4 Server, and the machine mentioned below. Someone else mentioned the same issue on a new Dual Xeon EM64T capable Tatung server, and some searching on their website shows TSS-2552 also uses the Intel E7520 Chipset Perhaps its just coincidence? Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sid Sent: Tuesday, 11 January 2005 7:35 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE410P problem (Looping UP Span 1...) Hi, This is the motherboard: SM X6DAE-XG2 Dual XEON 800FSB EMT64 w/2-Ch SATA-R 0&1,SVGA,2xGb LAN Dual Intel. Xeon EM64T Support up to 3.60 GHz Intel. E7520 (Lindenhurst) Chipset 1(x8) PCI-Express on (x16) Slot, 3 x 64-bit 133MHz PCI-X, 2 x 64-bit 100MHz PCI-X Slots ATI RageXL 8MB Graphics Actually it doesnt have 4 cpus as i mentioned in my earlier mail. It has 2 XEON cpus with hyper-threading technology. Anyone has seenexperienced any problems with this motherboard? -Sid --- Eric Bishop <[EMAIL PROTECTED]> wrote: > Is it a HP-Compaq DLXXX G4 machine? Because there is a thread here > about the TE410P not generating interrupts with these servers... > > > On Sat, 8 Jan 2005 05:14:39 -0800 (PST), Sid <[EMAIL PROTECTED]> wrote: > > Hi Scott, and Jack, > > > > --- Scott Stingel <[EMAIL PROTECTED]> wrote: > > > > > Sid- > > > > > > Try connecting one port to another. Note that one of the ports must be > > > set up as "cpe" and the other as "net" in zapata.conf when you loop them > > > together like this. > > > > > > A suitable crossover cable for testing can be constructed by cutting up > > > a CAT 5 cable, and connecting: > > > Pin 1 <--> Pin 4 on the other end > > > Pin 2 <--> Pin 5 > > > Pin 4 <--> Pin 1 > > > Pin 5 <--> Pin 2 > > > > > > You should get green's on both the connected channels if your zaptel and > > > zapata configurations are ok, and if you've run both modprobe and ztcfg > > > as documented. > > > > > > > Thanks for the valuable responses. We can only do the tests on monday, as the machine > is > > in a data center. Other than that we have done every tests we can think of and found > in > > the mailing list/wiki. The tests done at the NOC says that T1 is ok at their end. > Please > > see the following information about the system: > > > > The machine has 4 Xeon 2.80GHz CPUs. > > > > This is from /proc/interrupts > > 16: 0 0 0 3 IO-APIC-level usb-uhci > > 19: 0 0 0 0 IO-APIC-level usb-uhci > > 23: 0 0 0 0 IO-APIC-level ehci-hcd > > 26: 129024 0 0 28 IO-APIC-level eth1 > > 27: 0 0 21352 5 IO-APIC-level eth0 > > 76: 0 0 0 0 IO-APIC-level t4xxp > > > > # dmesg > > Zapata Telephony Interface Registered on major 196 > > Specify address with base=0xN > > Registered Tormenta2 PCI > > Found TE410P at base address fc8ff800, remapped to f8a40800 > > TE410P version c01a009b > > FALC version: 0005, Board ID: 00 > > Reg 0: 0x371c9800 > > Reg 1: 0x371c9000 > > Reg 2: 0x07fc07fc > > Reg 3: 0x > > Reg 4: 0x > > Reg 5: 0x > > Reg 6: 0xc01a009b > > Reg 7: 0x1000 > > Reg 8: 0x > > Reg 9: 0x00ff > > Reg 10: 0x > > TE410P: Launching card: 0 > > TE410P: Setting up global serial parameters > > Found a Wildcard: Wildcard TE410P-Xilinx > > Registered tone zone 0 (United States / North America) > > TE410P: Span 1 configured for ESF/B8ZS > > SPAN 1: Primary Sync Source > > > > I am doubtful about the interrupts. Are those values ok? We have been after this > problem > > for more than a week now, we have tested with 2 different cards to no success. > > > > I'll post the results of the crossover connection test, once we do that. Thanks again > for > > the responses. > > > > BR, > > -Sid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
On Sun, 2005-01-16 at 16:52 -0500, Steve Kann wrote: > If the delay goes down after a couple of minutes after the transfer, > this could be the problem. Just fyi, this is what I observed with those delays between iaxcomm and firefly, i.e. they occurred on a transfer attempt and normalized after some minutes of talking. Wouldn't be surprised if the transfer was the problem here, too. What I'm not sure about is, due to lack of thorough debugging, whether this is a * or iaxclient library issue ... Regards, Bruno. ___ 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] DIAX 0.9.9g more features and higher stability
On Jan 16, 2005, at 2:53 PM, Dan wrote: Hi Steve, - Original Message - From: "Steve Kann" <[EMAIL PROTECTED]> On Jan 14, 2005, at 2:03 PM, Dan wrote: Hi, \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > I dont have problems when calling PSTN extensions, and calling > VoceMail, EchoTest, etc. The problem is related with the conversation > between two DIAX Softphones. Between 2 DIAX phone and the delay is in one direction only?? Yes. One direction only... Just who make the call get the delay. Then try jitterbuffer=no in iax.conf to see if it solves this issue. Dan et. al, I think this might be a problem with native transfers, and needing to reset the jitterbuffer history when this happens, or something like this.. -SteveK But I have tried and I do don't have this problem here... What can I do to make this happen here? I don't know... Maybe if we could get a packet trace of the situation that causes the problem? Maybe try notransfer or whatever the iax.conf parameter is, and see if that changes things. If it does, it points towards this being the problem. If the delay goes down after a couple of minutes after the transfer, this could be the problem. If it doesn't, there's something else really wrong.. (I'm assuming you're using the new JB code here..). Also, if you're using the new JB code, you should implement the stuff to get the network stats, so we can see if calculated jitter is substantially higher..) ___ 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: RE: Re: [Asterisk-Users] Bristuff 0.20RC3 loses connectivity after short line interruption?
Tomorrow (monday) I will post my kernel oops messages together with my dmesg to Junghanns. I have noticed I cannot use the init.d script more then 1 or 2 times before the server dies completely. Prolly cause of the half unloaded module. Remco: I have simply connected the 2 NT1 boxes with a cat5 utp cable to the 2 HFC cards. There were cables in it that went into our very old Panasonic box. Simply unplugging those and repatching them on the pathpanel so they are directly connected did the trick here. We have been using this setup for a couple of business days now and all worked ok. Let me know if this works for you too. Michiel van Baak Terrazur - Originele Bericht - Van: Remco Barende Aan: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion Datum: Friday, 14 January 2005, 20:06 Onderwerp: RE: Re: [Asterisk-Users] Bristuff 0.20RC3 loses connectivity after short line interruption? Weird, I havent actually tried that, that may be part of my problem too. If i disconnect the line from the NT1 bristuff will not reconnect on every occasion. Disconnecting the cord between the Nt1 and the HFC-S card makes it lose connectivity occasionally. But I guess either way, the modules should restore the line connection. Does anybody know if any of the Junghanns developers follow this list or should we e-mail them a bug report? (I briefly looked through the tarball but didnt see an e-mail address but I didnt browse the sources). Groetjes, Remco P.S. Michiel : Do you have any experience connecting KPN PRI to *? I need to do that soon. On Fri, 14 Jan 2005, Michiel van Baak wrote: > We have 2 HFC-S cards. > We have a very simular problem here. > restarting * means no more outgoing calls. > I first have to unload the modules, load them again and start asterisk. > plugging/unplugging cables from the cards dont give any problems here. > I fixed the asterisk reloading thing by altering the /etc/init.d/asterisk > script. > > OT: I have the kernel oops messages when unloading the module like described > on voip-info.org > > Michiel van Baak > Terrazur > > >> - Originele Bericht - >> Van: Remco Barende >> Aan: Asterisk Users Mailing List - Non-Commercial Discussion Datum: >> Thursday, 13 January 2005, 18:39 Onderwerp: Re: [Asterisk-Users] Bristuff >> 0.20RC3 loses connectivity after short line interruption? >> >> Sorry I forgot to mention that. Its just a cheap ass HFC-S single BRI >> card (manufactured by E-Tech). I googled around and I know it can take >> some time to recover for the NT1 but I think this doesnt apply for s0. >> >> Even after waiting for 10 minutes I do not get any connectivity but >> unloading and reloading the modules seems to solve the problem instantly. >> >> I could even have a script do it as a really ugly way to solve it but I >> dont think there is any way for a script to know if the ISDN connection >> is lost or not. >> >> Remco >> >> >> On Thu, 13 Jan 2005, George Konstantoulakis wrote: >> >>> Same thing here, >>> I am using bristuff0.20-RC2b with an octoBRI card. >>> It only happens with DDI lines. With normal ISDN lines I dont have a >>> problem. >>> Which card are you using ? >>> >>> >>> Remco Barende wrote: >>> I installed bristuff0.20-RC3 (which includes * 1.0.3 stable) It works fine until I disconnect the phone jack for the ISDN line. Even after plug it back in asterisk still reports that it could not create a zap channel when I try to dial out and the line gives an engaged tone when I try to dial. Re-starting asterisk doesnt solve this, I have to stop asterisk, unload the modules, reload the modules and start asterisk again. I assume this is a bug, not a feature (should I e-mail it to Junghanns directly??)? I know the telco here in holland and I will lose the line for a short period every once in a while and its annoying when the line doesnt come back up. Or did I forget some setting to recover from such a situation? Thanks! ___ 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 >>> >> ___ >> 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] France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo
On Wed, 12 Jan 2005, Wilson Pickett wrote: Any chance to post a small how to with the correct server settings et al? I'm replying to the list, that what it's for. First of all, I think the big problem wasn't configuring asterisk but getting the username and password. To do this you'll need to set up their Windows SIP client and sniff the first connection to see how that's done. At some point you'll see a HTTP connection that requests your info via a PHP script. The return is a short XML that even I could figure out, something like wgyourpseudo refkrefre2gji4ih1yjdd9yhdgszfd6 Once you have that info, make sure you use callerid and configure the peer as usual. /hax0r n00b mode on Which command and parameters do I need to use to get some legible (usable) output to do the packet sniffing? I tried ethereal but it only gives me loads of garbage? /hax0r n00b mode off :) Thanks!! ___ 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] Guatemala DID's?
I'm looking for a company that offers Guatemala DID's. I saw that Lingo does, but Lingo isn't easily compatible w/ Asterisk, so they're a last resort. Thanks in advanced, Phil Astin. ___ 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] chan_sccp and bristuff 1.0.3 weirdness
I am using chan_sccp on bristuffed asterisk (0.2RC3 on asterisk 1.0.3).
Things seem fine but I am seeing some weird stuff. I have a Kirk IP600
connecting to * with 2 handsets.
The weird thing is that for incoming calls the handset that is put second
as my dialstring, never rings.
This is my dial string:
exten => 6,4,Dial(${PHONE1}&${DECT1}&${DECT2}),25,tm)
Where DECT1 & DECT2 are srings for SCCP/phonenr
If i specify DECT2 first and DECT1 second then DECT1 doesn't ring.
I did not see this behaviour on a non-bristuffed install of asterisk, both
phones worked as expected (same config files but on a box with an X100P).
The message on the console is :
Jan 16 22:10:15 NOTICE[5005]: chan_sccp.c:103 sccp_request: Can't find
SCCP/106): Unknown Line or Intercom
Is this a bug in chan_sccp or bristuff?
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Re: [Asterisk-Users] Status of latest round of Allison recordings
Are you almost done sorting the files? - Original Message - From: "Rob Fugina" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" ; "Asterisk Developers Mailing List" Sent: Thursday, January 13, 2005 12:19 PM Subject: [Asterisk-Users] Status of latest round of Allison recordings > Just wanted to catch everybody up. So many people contributed to the > work list that I'd probably miss several if I tried to notify just > them. Many people contributed funds, too, which I greatly appreciate. > We were able to get the entire list done, which added up to two full > hours of work on Allison's part. And that's just the recordings -- > I'm doing the editing myself. > > Which brings me to this: I'm just about finished with the editing. I > think it'll take me a couple more evenings to finish grouping and > naming all of the individual files appropriately, then I need to make > up the index, and I'll upload the whole thing to the bugtracker for > inclusion in CVS. This should all be done by a week from now. > > I did already upload prompts for several specific patches -- privacy > manager, meetme enhancements, phrase management... I think there was > one other. > > Attached is the final script Allison worked from, so everyone can see > what's on its way. > > Rob > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 lost after reboot
I had exactly the same issue with the newest card I got. I tried it with Zaptel drivers from CVS HEAD and the problem disappeared. It could be that older drivers don't work with the latest cards. Mark wrote: Do you have your zaptel drivers set to start when the system is rebooted? If not, try rebooting and issue the "modprobe zaptel" and "modprobe wctdm" commands to manually start them. You could also issue the "lsmod" command after a reboot to see if zaptel and wctdm are running. I had problems with the zaptel startup script, but for whatever reason it works now. Good luck! Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 16, 2005 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400 lost after reboot Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ 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 ___ 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] TE410P card in an HP-Compaq DL380 G4 server
On Sat, 15 Jan 2005, Begumisa Gerald M wrote: > > Yup, I found their support very unhelpful and unwilling to go the > > extra (or even the first) mile.. > > Might ACPI (not APIC) have anything to do with this condition? I once had > a hard time with a bunch of cards which were not taking interrupts. I > disabled ACPI interrupt routing (from the grub boot prompt, put > pci=noacpi) and everything started working. Well, these were TDM400P > cards (5 of them) anyway with a different type of machine altogether but > it just might be worth checking out. I did try pci=noacpi and also compiling the kernel with it turned off - both to no avail. Steve ___ 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] TE410P card in an HP-Compaq DL380 G4 server
On Fri, 14 Jan 2005, Steve Hanselman wrote: > Has anyone also logged a support call with Digium, it has to be either the > card, Linux or the Zaptel drivers. > Yes of course - we have a call open. Steve ___ 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] Type of Number
On Sun, 16 Jan 2005, Marc Storck wrote:
> how can I read the PRI type of number:
>
> [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
> E.164/E.163) (1)
> < Presentation: Presentation allowed of network provided number (3)
> '061706161' ]
>
> (in this case TON = 2)
>
> Does a variable like ${TON} exist??? Or how can i read that number?
>
> If this would have to be implemented I'm willing to fund a bounty!
CALLERTON should have held that value. Unfortunatly, it does not work for
pri channels. I have a fix but I am still waiting for the legal
department to sign the Digium disclaimer. I can send it to you for testing
though. Actually, that would probably make the acutal bug report easier
once the disclaimer is signed. Contact me if you want to test the patch.
Peter
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[Asterisk-Users] Type of Number
Hello,
how can I read the PRI type of number:
[ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
E.164/E.163) (1)
< Presentation: Presentation allowed of network provided number (3)
'061706161' ]
(in this case TON = 2)
Does a variable like ${TON} exist??? Or how can i read that number?
If this would have to be implemented I'm willing to fund a bounty!
Regards,
Marc
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
Internet Service Provider http://www.msnetworks.lu
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---
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Re: [Asterisk-Users] kind of urgent
I had the same issue. did you ever find a solution. The Fritz card worked fine with FC2, but no go with FC3, I think it has to do with udev. On Thu, 06 Jan 2005 19:36:17 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote: > > Though you probably won't use them, I'd still like to mention fyi that > proprietary AVM Fritz PCI Card drivers didn't work for me on FC3. They > did on Debian Sarge. > > Regards, Bruno. > > > On Thu, 2005-01-06 at 19:32 +0200, Shoval Tomer wrote: > > Hi all. > > > > Can anyone comment why shouldn't we use FC 3 for an * production system? > > > > I'm not looking to start a distro war, but we just found out that redhat > > 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. > > > > We are only familiar with red hat and are in a point in time that > > switching distros is not available. > > The guy installing the system is already on location. > > > > Yes, I know we made a silly mistake. Please help us... > > Thanks. > > > > Shoval > > > > > > ___ > > 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 > ___ 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] Inbound Callerid for SIP Phones
I have a number of inbound analog lines connecting through Digium cards to an Asterisk box. Asterisk then bridges the calls over to the internal extensions which are all SIP phones. Currenly the inbound lines do not have callerid on them so callerid=no in my zapata.conf file. What happens on inbound calls is that the SIP extensions are dialed but their callerid shows '[EMAIL PROTECTED]:X.com'. Does anyone know how to change the callerid on the inbound calls? I would like to change it to something like Inboud Call or something more descricptive than asterisk. ___ 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] DIAX 0.9.9g more features and higher stability
Hi Steve, - Original Message - From: "Steve Kann" <[EMAIL PROTECTED]> On Jan 14, 2005, at 2:03 PM, Dan wrote: Hi, \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > I dont have problems when calling PSTN extensions, and calling > VoceMail, EchoTest, etc. The problem is related with the conversation > between two DIAX Softphones. Between 2 DIAX phone and the delay is in one direction only?? Yes. One direction only... Just who make the call get the delay. Then try jitterbuffer=no in iax.conf to see if it solves this issue. Dan et. al, I think this might be a problem with native transfers, and needing to reset the jitterbuffer history when this happens, or something like this.. -SteveK But I have tried and I do don't have this problem here... What can I do to make this happen here? Best regards, Dan ___ 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] CAC Channel Bank I - FXS
I solved this issue. DIP switches marked Option A & Option B need to be off (down). -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Richard Cook > Sent: Sunday, January 16, 2005 1:56 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Cc: 'Andrew Kohlsmith' > Subject: RE: [Asterisk-Users] CAC Channel Bank I - FXS > > > Subject: Re: [Asterisk-Users] CAC Channel Bank I - FXS > > > > On January 15, 2005 03:39 pm, [EMAIL PROTECTED] wrote: > > > I have a CAC Channel Bank I with FXS cards. I've the > system up and > > > running, with just 1 issue. > > > > > When I make an inbound call, Asterisk says "Zap/26 is ringing", > > > however, the phone never rings. No lights are lit on the > > CAC during the calll. > > > > None at all, or just no change from idle? If you pick up the line > > that you're ringing (but not hearing it ring), does it connect your > > call? > > On idle, I have no lights on at all. When the channel rings, > no lights are lit. If I pick up the line while it's ringing, > yes, it does connect my call. > > The only time a light is lit, is when I pick up the receiver > -- it goes red (in use -- outgoing). > > > > Outbound call works no problem, and the CAC lights up correctly. > > > Any ideas what could be the problem? > > > > We could be of far more assistance if you posted the relevant bits > > from your zapata.conf and your zaptel.conf files... > > Here is my zaptel.conf: > > # span 1 = PRI > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > # span 2 = Channel Bank 1 > span=2,0,0,esf,b8zs > fxols=25-48 > # span 3 = Spare > span=3,0,0,esf,b8zs > # span 4 = Spare > span=4,0,0,esf,b8zs > # > unused=49-96 > > Here is my zapata.conf: > > ; Channel Bank - port 1 - Sivana/VocTel alarm ; > signalling=fxo_ls context=internal > callerid=7054979051 > channel=>25 > ; > ; Channel Bank - port 2 - Sivana/VocTel Fax ; > signalling=fxo_ls context=internal > callerid=7054979051 > channel=>26 > > I'm trying to call channel 26 which is the fax line (with a > handset plugged in). > > -- > Richard Cook > [EMAIL PROTECTED] > Tel: 705-497-9320 ext 2010 > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
On Jan 14, 2005, at 2:03 PM, Dan wrote: Hi, \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > I dont have problems when calling PSTN extensions, and calling > VoceMail, EchoTest, etc. The problem is related with the conversation > between two DIAX Softphones. Between 2 DIAX phone and the delay is in one direction only?? Yes. One direction only... Just who make the call get the delay. Then try jitterbuffer=no in iax.conf to see if it solves this issue. Dan et. al, I think this might be a problem with native transfers, and needing to reset the jitterbuffer history when this happens, or something like this.. -SteveK ___ 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] MVP110 and *
Hi all, I am sure there is a way to get a Multitech MVP110 working with * in H.323 mode. I have just not been able to figure out how from the MVP110 side. Could someone please share their config setup with me for the MVP110 and the * side?? TIA, Robert Webb ___ 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] TDM400 lost after reboot
Do you have your zaptel drivers set to start when the system is rebooted? If not, try rebooting and issue the "modprobe zaptel" and "modprobe wctdm" commands to manually start them. You could also issue the "lsmod" command after a reboot to see if zaptel and wctdm are running. I had problems with the zaptel startup script, but for whatever reason it works now. Good luck! Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 16, 2005 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400 lost after reboot Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ 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
[Asterisk-Users] H323 Softphone for iPAQ
Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid ___ 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] CAC Channel Bank I - FXS
> Subject: Re: [Asterisk-Users] CAC Channel Bank I - FXS > > On January 15, 2005 03:39 pm, [EMAIL PROTECTED] wrote: > > I have a CAC Channel Bank I with FXS cards. I've the system up and > > running, with just 1 issue. > > > When I make an inbound call, Asterisk says "Zap/26 is ringing", > > however, the phone never rings. No lights are lit on the > CAC during the calll. > > None at all, or just no change from idle? If you pick up the > line that you're ringing (but not hearing it ring), does it > connect your call? On idle, I have no lights on at all. When the channel rings, no lights are lit. If I pick up the line while it's ringing, yes, it does connect my call. The only time a light is lit, is when I pick up the receiver -- it goes red (in use -- outgoing). > > Outbound call works no problem, and the CAC lights up correctly. > > Any ideas what could be the problem? > > We could be of far more assistance if you posted the relevant > bits from your zapata.conf and your zaptel.conf files... Here is my zaptel.conf: # span 1 = PRI span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # span 2 = Channel Bank 1 span=2,0,0,esf,b8zs fxols=25-48 # span 3 = Spare span=3,0,0,esf,b8zs # span 4 = Spare span=4,0,0,esf,b8zs # unused=49-96 Here is my zapata.conf: ; Channel Bank - port 1 - Sivana/VocTel alarm ; signalling=fxo_ls context=internal callerid=7054979051 channel=>25 ; ; Channel Bank - port 2 - Sivana/VocTel Fax ; signalling=fxo_ls context=internal callerid=7054979051 channel=>26 I'm trying to call channel 26 which is the fax line (with a handset plugged in). -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 ___ 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] The BEST? analog phones for *
Googling the archives there is some debate about what are good analog phones to use with *. Aastra seems popular, but they are somewhat pricey and the proprietary seems like it can be a headache. Can someone weigh in on what would be good analog phones for a small office (8 lines and 20 phones) to use with *. So far I'm most impressed with Smartalk primarily bc. they're don't use ADSI, they look nice and they seem somewhat reasonably priced. Anyone have any relevant experiences? Thanks, Richard __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] TDM400p FXS not sending caller id info?
Just in case anyone else has this problem, I'll list my solution: The latest CVS stable version (either zaptel or asterisk CVS) seemed to be the problem. When I installed 1.0.3, everything worked. matt Matthew Henkler wrote: I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few analog caller id devices, and none seem to be picking it up. I'd really like to be able to use my analog cordless handset and be able to see the caller id information. Zaptel config files below. Asterisk and zaptel are all latest 1.0 stable releases. Thanks! matt ___ 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] announcing caller id?
sorry I copied and pasted from the already posted stuff it should read:
You can try this:
exten => 101/,1,Dial(device,options...)
exten => 101,1,Dial(device,M(acallerid))
exten => 101,2,Voicemail(u${EXTEN})
exten => 101,102,Voicemail(b${EXTEN})
[macro-acallerid]
;assuming that:
; incoming.gsm exists and says:
; You have an incming call from..
; and options.gsm exists and says:
; to accept press 1, to send to voice mail press 2.
exten => s,1,Playback(incoming)
exten => s,2,Saydigits(${CALLERIDNUM})
exten => s,3,Read(ACCEPT|options|1)
exten => s,4,Gotoif($[${ACCEPT} = 1] ?50) ;connect
exten => s,5,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
exten => s,30,SetVar(MACRO_RESULT=CONTINUE)
exten => s,31,Goto(50)
exten => s,50,Noop("")
You can follow the following instructions to do more:
http://lists.digium.com/pipermail/asterisk-users/2004-December/080417.html
On Sun, 16 Jan 2005 13:15:12 -0500, C F <[EMAIL PROTECTED]> wrote:
> You can try this:
> exten => 101/,1,Dial(device,options...)
> exten => 101,1,Dial(device,M(acallerid))
> exten => 101,2,Voicemail(u${EXTEN})
> exten => 101,102,Voicemail(b${EXTEN})
>
> [macro-acallerid]
> ;assuming that:
> ; incoming.gsm exists and says:
> ; You have an incming call from..
> ; and options.gsm exists and says:
> ; to accept press 1, to send to voice mail press 2.
>
> exten => s,1,Playback(incoming)
> exten => s,2,Saydigits(${CALLERIDNUM})
> exten => s,3,Background(options)
> exten => s,4,Read(ACCEPT|custom/screnn-accept|1)
> exten => s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
> exten => s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
>
> exten => s,30,SetVar(MACRO_RESULT=CONTINUE)
> exten => s,31,Goto(50)
> exten => s,50,Noop("")
> You can follow the following instructions to do more:
> http://lists.digium.com/pipermail/asterisk-users/2004-December/080417.html
>
>
> On Sun, 16 Jan 2005 18:01:11 +0100, Dave Cotton
> <[EMAIL PROTECTED]> wrote:
> > On Sun, 2005-01-16 at 14:49 +0100, Robert Rozman wrote:
> >
> > > could you please give some more info how to do this ?
> >
> > Use Custom ring 1 tone with with a blank Caller ID
> >
> > --
> > Dave Cotton <[EMAIL PROTECTED]>
> >
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [Asterisk-Users] announcing caller id?
You can try this:
exten => 101/,1,Dial(device,options...)
exten => 101,1,Dial(device,M(acallerid))
exten => 101,2,Voicemail(u${EXTEN})
exten => 101,102,Voicemail(b${EXTEN})
[macro-acallerid]
;assuming that:
; incoming.gsm exists and says:
; You have an incming call from..
; and options.gsm exists and says:
; to accept press 1, to send to voice mail press 2.
exten => s,1,Playback(incoming)
exten => s,2,Saydigits(${CALLERIDNUM})
exten => s,3,Background(options)
exten => s,4,Read(ACCEPT|custom/screnn-accept|1)
exten => s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
exten => s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
exten => s,30,SetVar(MACRO_RESULT=CONTINUE)
exten => s,31,Goto(50)
exten => s,50,Noop("")
You can follow the following instructions to do more:
http://lists.digium.com/pipermail/asterisk-users/2004-December/080417.html
On Sun, 16 Jan 2005 18:01:11 +0100, Dave Cotton
<[EMAIL PROTECTED]> wrote:
> On Sun, 2005-01-16 at 14:49 +0100, Robert Rozman wrote:
>
> > could you please give some more info how to do this ?
>
> Use Custom ring 1 tone with with a blank Caller ID
>
> --
> Dave Cotton <[EMAIL PROTECTED]>
>
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[Asterisk-Users] No compatible codecs
I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: -- Accepting AUTHENTICATED call from 192.168.112.99, requested format = 512, actual format = 512 -- Called [EMAIL PROTECTED] -- SIP/mutualphone-6b26 is ringing -- SIP/mutualphone-6b26 answered IAX2/[EMAIL PROTECTED]/2 The BT101 gives this: -- Called [EMAIL PROTECTED] -- SIP/mutualphone-2de1 is ringing -- SIP/mutualphone-2de1 answered SIP/chimit01-6013 -- Attempting native bridge of SIP/chimit01-6013 and SIP/mutualphone-2de1 Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No compatible codecs! -- Got SIP response 488 "Not Acceptable Here" back from 209.250.147.116 show translation (I figure this has anything to do with it) shows that all paths are supported: G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 4 2 2 3 2 1 4133519 GSM15 - 2 2 3 2 1 4133519 ULAW15 4 - 1 3 2 1 4133519 ALAW15 4 1 - 3 2 1 4133519 G72617 6 4 4 - 4 3 6153721 ADPCM15 4 2 2 3 - 1 4133519 SLINR14 3 1 1 2 1 - 3123418 LPC1017 6 4 4 5 4 3 -153721 G729A17 6 4 4 5 4 3 6 -3721 SPEEX16 5 3 3 4 3 2 514 -20 ILBC17 6 4 4 5 4 3 61537 - The first preferred Vocoder configured in the BT101 is PCMU, but changing this to G729 (the one that mutualphone is using) won't make it work. I changed the option back again because all other services (FWD, BRI, IAX2) work like this and I don't want to break them. Any suggestions about what I can change to make this work? Cheers! Rene Kluwen Chimit ___ 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] VoIP Newbie
Hello All, I am trying very hard to learn what it takes to set up an VoIP service that will allow users to download some comunications software possibly like "SIP Communicator" or something better and using the Asterisk PBX software. The problem is that I am a little confused as to what I all I need. I have a lot of Linux/Windows development and networking experience but almost zero VoIP experience. The goal is to set up a small Internet software phone VoIP service with video/audio and then expand to add standard PBX hard-lines as well to a local area. Any advice would be greatly appreciated, Thanks, Lonnie ___ 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] NuFone help
Jake Franklin wrote:
Hello,
I've signed up for a NuFone account, and added the following
instructions to my config files per NufFones directinos:
iax.conf
[NuFone]
type=peer
host=switch-1.nufone.net
secret=password
extensions.conf
(under the [default] context)
exten => _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I then get this message in the CLI:
-- Executing Dial("SIP/jake-fe5d",
"IAX2/user:[EMAIL PROTECTED]/1303555") in new stack
-- Called user:[EMAIL PROTECTED]/1303555
-- Call accepted by 66.225.202.72 (format gsm)
-- Format for call is gsm
-- Hungup 'IAX2/NuFone/1'
== No one is available to answer at this time
I have, of course, changed the username/passwd and phone # for
security reasons in this e-mail.
Any help would be greatly appreciated!
Jake
i had this issue the other day. and a cvs-head update fixed it. (there
was a kernel update fix that happened right around the same time and i'm
not completely sure which fixed it (hense the mentioning of it))
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[Asterisk-Users] Re: asterisk-users list and html posts
Sorry about the HTML post, I was sending from my laptop and forgot to turn off html in outlook. Have a nice day. -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Sunday, January 16, 2005 12:50 AM To: [EMAIL PROTECTED] Subject: Re: asterisk-users list and html posts Please do not post to the list with HTML or the stationary crap... its really annoying. Thanks, bkw_ ___ 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] Looking for help with a Polycom Soundpoint IP 600
Aryeh wrote: Hi all, I have a Polycom Soundpoint IP 600 that looks like it is fried. It either has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt that asks if I want to enter the setup configuration or continue booting. When plugged into power, it turns on, shows the Polycom logo for 3 seconds and then pretty much goes dead with two red lights on. The one in the top right corner and the one in the top left (the line 1 red light). Anyone have any clues as to what to do? I'm looking into finding a polycom service center or reseller that is in or close to New York City. Anyone know of any places that can simply connect up to the phone and reflash the rom and firmware. I'm guessing it's a 5 min fix with the right equipment. Any help would be appreciated. Thanks guys. aryeh ___ 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 Aryeh - We are a Polycom authorized reseller, we may be able to help you get your IP600 repaired with Polycom, depending on when you purchased it. If you like, please send me the serial number off the unit, as well as the approximate date you purchased it, and I will look into it for you. Thanks Cory @ VOIPSupply.com ___ 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] announcing caller id?
On Sun, 2005-01-16 at 14:49 +0100, Robert Rozman wrote: > could you please give some more info how to do this ? Use Custom ring 1 tone with with a blank Caller ID -- Dave Cotton <[EMAIL PROTECTED]> ___ 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] Looking for help with a Polycom Soundpoint IP 600
Hi all, I have a Polycom Soundpoint IP 600 that looks like it is fried. It either has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt that asks if I want to enter the setup configuration or continue booting. When plugged into power, it turns on, shows the Polycom logo for 3 seconds and then pretty much goes dead with two red lights on. The one in the top right corner and the one in the top left (the line 1 red light). Anyone have any clues as to what to do? I'm looking into finding a polycom service center or reseller that is in or close to New York City. Anyone know of any places that can simply connect up to the phone and reflash the rom and firmware. I'm guessing it's a 5 min fix with the right equipment. Any help would be appreciated. Thanks guys. aryeh ___ 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] IAX2 one side loses audio
I'm experiencing a similar issue, but it's with IAX2 / IAX2 calls. I've started to think that it's a router or something upstream. For me, if I keep the call bridged through asterisk (notransfer=yes), after about a minute of conversation, the called party can't hear the caller. I watched the traffic coming out of asterisk using iptraf, and it doesn't seem to change when they lose audio. It is really frustrating for us, as we've already been having to work out issues with our ISP on their QoS settings. Ron Senykoff Systems Architect / Developer HarrisLogic Inc. 972-215-0488 x 3020 312-404-8745 (cell) Trevor Peirce <[EMAIL PROTECTED] on.ca> To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] m.com cc Subject 01/15/2005 06:28 [Asterisk-Users] IAX2 one side PMloses audio Please respond to Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED] ists.digium.com> It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists. Thanks, Trevor Peirce ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] announcing caller id?
Chris Polk wrote: Any one have any solution for this? We need to have the caller id information announced when the phone is answered. for example I am sitting at my desk, my phone rings. I pick it up and hear call from 55 to except press 1 to decline press to any help would be grately appreciated! "show application dial" Pay special attention to the M() option. ___ 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] Re: Asterisk-Users Digest, Vol 6, Issue 227
Thanks! Thanks! Thanks! I've got it work!!! :-) Message: 13 Date: Sun, 16 Jan 2005 12:17:21 - From: "Bill Seddon" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] failed to compile zaptel on redhat To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" << then it looks like the does not have the full kernel sources installed>> ...or isn't running the 2.6 kernel. I had the same problem with the CVS on Friday (but not from the week before). It turns out that moduleparam.h is included as part of a bug fix on 2.6 but instead of being #ifdef'd for 2.6 and later, the inclusion was absolute causing compilations of zaptel on earlier Linux kernels to fail. The advice I received was: The culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include to: #ifdef LINUX26 #include #endif Bill Seddon __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ 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] TDM400P NO BATTERY & Poopy???
i just checked out the digium site, but its a bit expensive and i'll end up w/ two fxo modules that i'll never need. if anybody would be interested in swapping two fxs modules for two fxo's it would be a great help, please contact me offlist. thanks, jon - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 16, 2005 10:30 AM Subject: Re: [Asterisk-Users] TDM400P NO BATTERY & Poopy??? doh! i assumed the x100p and TDM400p worked the same, because i thought was able to do both on that card...well thanks for the help :( Side note : you just have to get 2 FXS modules for your TDM400, the card can use FXO or FXS modules, and you can mix them as you wish ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P NO BATTERY & Poopy???
> doh! i assumed the x100p and TDM400p worked the same, because i thought was > able to do both on that card...well thanks for the help :( Side note : you just have to get 2 FXS modules for your TDM400, the card can use FXO or FXS modules, and you can mix them as you wish ___ 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] TDM400P NO BATTERY & Poopy???
doh! i assumed the x100p and TDM400p worked the same, because i thought was
able to do both on that card...well thanks for the help :(
-jon
- Original Message -
From: Henry Devito
To: Asterisk-Users@lists.digium.com ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, January 16, 2005 1:24 AM
Subject: Re: [Asterisk-Users] TDM400P NO BATTERY & Poopy???
That's because to hook analog phones up to a port the port must be fxs. So
for your situation you need a card with 2 FXO for CO lines and 2 FXS for
regular phones.
---Original Message---
From: [EMAIL PROTECTED]
Date: 01/16/05 00:16:21
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TDM400P NO BATTERY & Poopy???
All,
I've recently installed a TDM400P and am unable to get it working. It has
four FXO modules on it. What I would like to do is have the first two ports
as inbound from pstn and have them ring the last two ports which will be
connected to regular analog phones.
In Zaptel.conf I have:
fxsks=1-4
In Zaptel.conf
[channels]
context=default
switchtype=national
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
sendcalleridafter=1
callwaitingcallerid=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
immediate=no
signalling=fxs_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
group=1
channel =>1-4 ; Again X is the number of FXO modules you have
In extensions.conf
exten =>_NXX,1,Dial(Zap/g1/${EXTEN}|20,t)
exten =>s,1,Wait(1)
exten =>s,2,Dial,Zap/g1
exten =>s,3,Voicemail,u9000
exten =>s,4,Hangup
I've also tried to see the debugging output, and I get the following errors:
Can someone point me in the right direction.
TIA,
Jon
[EMAIL PROTECTED] asterisk]#
[EMAIL PROTECTED] asterisk]# /sbin/ztcfg
[EMAIL PROTECTED] asterisk]# tail -f /var/log/messages
Jan 16 00:59:38 localhost kernel: ISO-Cap is now up, line side: 03 rev 03
Jan 16 00:59:38 localhost kernel: Module 3: Installed -- AUTO FXO (FCC mode)
Jan 16 00:59:38 localhost kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F
(4 modules)
Jan 16 00:59:38 localhost kernel: NO BATTERY on 1/2!
Jan 16 00:59:38 localhost kernel: NO BATTERY on 1/3!
Jan 16 00:59:38 localhost kernel: NO BATTERY on 1/4!
Jan 16 00:59:38 localhost kernel: 213781 Polarity reversed (0 -> 1)
Jan 16 00:59:50 localhost kernel: Registered tone zone 0 (United States /
North America)
Jan 16 01:00:03 localhost kernel: Poopy (00) on card 1!
Jan 16 01:00:03 localhost kernel: Poopy () on card 1!
Jan 16 01:00:05 localhost last message repeated 2 times
Jan 16 01:00:12 localhost kernel: Poopy (00) on card 1!
Jan 16 01:00:12 localhost kernel: Poopy () on card 1!
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Re: [Asterisk-Users] NuFone help
try iax2 debug
> Hello,
>
> I've signed up for a NuFone account, and added the following
> instructions to my config files per NufFones directinos:
>
> iax.conf
> [NuFone]
> type=peer
> host=switch-1.nufone.net
> secret=password
>
> extensions.conf
> (under the [default] context)
> exten => _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
>
> I then get this message in the CLI:
>
> -- Executing Dial("SIP/jake-fe5d",
> "IAX2/user:[EMAIL PROTECTED]/1303555") in new stack
> -- Called user:[EMAIL PROTECTED]/1303555
> -- Call accepted by 66.225.202.72 (format gsm)
> -- Format for call is gsm
> -- Hungup 'IAX2/NuFone/1'
>== No one is available to answer at this time
>
> I have, of course, changed the username/passwd and phone # for security
> reasons in this e-mail.
>
> Any help would be greatly appreciated!
>
> Jake
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[Asterisk-Users] * reports the incoming caller id but not the BT100
On incoming calls it seems that * is finding the callerid correctly but my BudgeTone is not showing it in the display. What am I doing wrong? The * console shows: -- Accepting call from '6' to '6' on channel 0/1, span 1 (numbers changed) but I guess that'c correct? ___ 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] TDM400 lost after reboot
Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ 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] announcing caller id?
Hi, - Original Message - From: "Wilson Pickett" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 16, 2005 10:20 AM Subject: Re: [Asterisk-Users] announcing caller id? > > Any one have any solution for this? > > We need to have the caller id information announced when the phone is > > answered. > > for example > > I am sitting at my desk, my phone rings. > > I pick it up and hear call from 55 to except press 1 to decline > > The Grandstream BT100 series phones will do this without the help of > asterisk. Otherwise, you can write an extension to do it, allowing for > calls with callerid disabled of course. could you please give some more info how to do this ? Thanks, Rob. > ,___ > 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
[Asterisk-Users] TDD support in Asterisk?
I can see that Asterisk supports TDD text channels, as there is a tdd.c file in the source, this appears to be exposed via TDD MODE in the AGI interface? I cant find any documentation anywhere on how to use this. Has anyone done this? tdd.c seems to only support 45.5 baud calls. Ideally I'd like a simple TDD answering machine, that sends "PLEASE TYPE MESSAGE" and receives TDD calls via SIP then sends me the text in an email. I appreciate this is a bit of an unusual query... Andy ___ 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] Re: SS7 and Asterisk solution
Erm, at the risk of getting flamed, where does IAX come into this picture? If I re-implement IAX(2) in a different language (not using iaxcomm except as a refererence or test ) and want to sell a product based on it can I do that, or do I need a license ? You are probably ok without a comercial license. It depends on how heavily you borrow directly from the gpl:ed source. The usual (unclear to me) rules for what constitutes a "derived work" applies. If you start from the specification there should be no problem whatsoever. When in doubt contact the original author and/or your legal councel. You can modify and/or link to GPLed code with commercial code and get away with it as long as you don't distribute the stuff. That's the story with G.729, with nVidia drivers etc etc etc roy ___ 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] ATA186: SIP/2.0 503 Service Unavailable
> >>I have done my homework on this, I hope.
> >>
> >>I have a customer with an ATA186 who uses Nufone as his IAX provider.
> >>His network operations center in the Bahamas was destroyed by the
> >>hurricanes, and I'm helping him rebuild.
> >
> >
> > I can help, but I think it might require being on site.
> >
> > Just kidding; its 9 degrees above zero here in Nebraska. :(
> >
> > Will need a little bit more then what you've provided to even guess
> > at the issue.
> >
> > Have you executed a 'sip debug' and looked at the detail?
> >
>
> It took me a while to get it sanitized--it's at a customer site. No NAT
> anywhere, 1.2.3.4 and 1.2.3.41 are the Asterisk box and ATA186,
> respectively. 81 is the "dial prefix" to choose the carrier. Also,
> iaxy calls in the same context, using the same exact dialstring, go out
> just fine. . .*very perplexing.*
>
> Thx.
>
> B.
>
> Snip
>
> hostname-II*CLI> sip debug
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 1.2.3.41:5060
> From: sip:[EMAIL PROTECTED];tag=2980654425
> To:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> Contact:
> User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
> Expires: 300
> Content-Length: 246
> Content-Type: application/sdp
>
> v=0
> o=ata7001 6010 6010 IN IP4 1.2.3.41
> s=ATA186 Call
> c=IN IP4 1.2.3.41
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Using latest request as basis request
> Sending to 1.2.3.41 : 5060 (non-NAT)
> Found RTP audio format 0
> Found RTP audio format 4
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port 1.2.3.41:16384
> Found description format PCMU
> Found description format G723
> Found description format PCMA
> Found description format telephone-event
> Capabilities: us - 0x4(ULAW), peer -
> audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
> Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
> 0x1(G723)
> Reliably Transmitting (no NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 1.2.3.41:5060
> From: sip:[EMAIL PROTECTED];tag=2980654425
> To: ;tag=as5307f0b3
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="5e9f7505"
> Content-Length: 0
>
>
> to 1.2.3.41:5060
> Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
> Found user 'ata7001'
>
> Sip read:
> ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 1.2.3.41:5060
> From: sip:[EMAIL PROTECTED];tag=2980654425
> To: ;tag=as5307f0b3
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 ACK
> User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
>
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 1.2.3.41:5060
> From: sip:[EMAIL PROTECTED];tag=2980654425
> To:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> Contact:
> User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
> Proxy-Authorization: Digest
>
username="ata7001",realm="asterisk",nonce="5e9f7505",uri="sip:[EMAIL
PROTECTED]",response="21
680b72deb8cb966868d671528fc431"
> Expires: 300> sip no debug
> Content-Length: 246
> Content-Type: application/sdp
>
> v=0
> o=ata7001 6016 6016 IN IP4 1.2.3.41
> s=ATA186 Call
> c=IN IP4 1.2.3.41
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 12 headers, 11 lines
> Using latest request as basis request
> Sending to 1.2.3.41 : 5060 (non-NAT)
> Found RTP audio format 0
> Found RTP audio format 4
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port 1.2.3.41:16384
> Found description format PCMU
> Found description format G723
> Found description format PCMA
> Found description format telephone-event
> Capabilities: us - 0x4(ULAW), peer -
> audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
> Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
> 0x1(G723)
> Found user 'ata7001'
> Looking for 811235551212 in home
> list_route: hop:
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 1.2.3.41:5060
> From: sip:[EMAIL PROTECTED];tag=2980654425
> To: ;tag=as29aecdb3
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Content-Length: 0
>
>
> to 1.2.3.41:5060
> -- Executing Dial("SIP/ata7001-76d6",
> "IAX2/[EMAIL PROTECTED]/11235551212") in new stack
> -- Called [EMAIL PROTECTED]/11235551212
> -- Call accepted by 66.225.202.72 (format ULAW)
> -- Format for call is ULAW
> -- Hungup 'IAX2/NuFone/7'
>== No one is available to answer
RE: [Asterisk-Users] Zaptel in HEAD broken?
Soren, thanks for the information and advice. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: January 14, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel in HEAD broken? Bill Seddon wrote: > Are there new instructions for compiling the zaptel driver in HEAD? > > I compiled the zaptel driver from HEAD successfully last weekend but > trying to compile the current driver for another machine results in > the error: > > zaptel.c:45:31: linux/moduleparam.h: No such file or directory > > If I go to compile zaptel on the machine that compiled successfully > last weekend, the same error occurs. So far as I can tell, I don't > have a file called moduleparam.h anywhere on either machine. Yeah, the culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include to: #ifdef LINUX26 #include #endif and now it compiles on 2.4... /Soren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] failed to compile zaptel on redhat
<< then it looks like the does not have the full kernel sources installed>> ...or isn't running the 2.6 kernel. I had the same problem with the CVS on Friday (but not from the week before). It turns out that moduleparam.h is included as part of a bug fix on 2.6 but instead of being #ifdef'd for 2.6 and later, the inclusion was absolute causing compilations of zaptel on earlier Linux kernels to fail. The advice I received was: The culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include to: #ifdef LINUX26 #include #endif Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard Sent: January 16, 2005 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] failed to compile zaptel on redhat On Sunday 16 January 2005 04:29, Steven Critchfield wrote: > linux/moduleparam.h is actually part of the kernel source. It is created > when you config and compile the kernel. It holds the version symbols > needed to properly link the new drivers into the kernel. No, it is part of the virgin kernel sources and defines the kernel modules parameters api. If he does not have it, then it looks like he does not have the full kernel sources installed. > I suggest you find a kernel compile howto that is at least understanding > of anything specific to the brokenness of RedHat and follow the > suggestions found within. >[...] > > Xu, Duo wrote: > > > why linux/moduleparam.h is missing in the source? I > > > saw it in 2.6 source. B ___ 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
[Asterisk-Users] sound-recorder crash when I start Asterisk
My problem used to come and goe without knowing the cause and the remedy .But now it is consistent. Each time I boot my Redhat9, I get the error message ["gnome-sound-recorder" (process #) has crashed due to fatal error (Aborted)]. After I close the error window and open the application "Sound Recorder" from the start menu, I am able to record my voice and play it back. When I make a third party VOIP call, I can hear the person I am calling, but he can't hear me. When I go back and check the "Sound Recorder", I can't record and play my voice. The application crashes and I can't close the window without rebooting the system. I greatly appreciate any help, I am still a newbie to both linux and Asterisk __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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] Extension.conf, sip.conf and contexts.
Hi, Does the context defined in sip.conf have to be the same context to which the extension belongs to in sip.conf? I have all my local SIP phones in context=local, and are in a context call local in extensions.conf. I then signed up with a few voip providers and I only wanted to allow one of the SIP phones to use it, so I moved him into his own context in both files. I also included [local] in entensions.conf. He can call all the other phones, and is able to call out using the voip providers, but the other SIP phones are unable to call him. Am I missing something very obvious? Regards David Norton ___ 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] IAX2 Channels & Bandwidth
Hi - I found the trunkfreq directive in iax.conf so I've put the directive line into the iax peers section (along with "trunk=yes") - I'm sure you meant iax.conf rather than zapata.conf ? Thanks for the help, Derek -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 16 January 2005 00:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Channels & Bandwidth On January 15, 2005 03:11 pm, Leif Madsen wrote: > This is true if you are using IAX2 trunking. This can be enabled with > trunk=yes in your peer configuration. The other end must also > support the trunking as well. Also if you're using iLBC you need to set the trunking period to 30ms instead of the default 20. trunkfreq=30 in zapata.conf should do it. -A. ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.809 / Virus Database: 551 - Release Date: 09/12/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.809 / Virus Database: 551 - Release Date: 09/12/2004 ___ 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] Security audit scripts
On Fri, 14 Jan 2005, Rich Adamson wrote: Are there security concerns with the * application software? I know there are with the Linux installation. :-) You should always be concerned with security. Not to say that Asterisk has any security problems (it is audited regularly). If you are administering network boxes you should really read up on network security. That said, most of your security concerns are going to come from applications which are running by default on your distro. You should really go through every application running on your box and decide a) whether you need it and b) what settings you really need. This has sort of been discussed before on the list, but I'd suggest there is a much larger security issue running asterisk resulting from the implementor not understanding "contexts". I'm not talking about problems with the code, but rather experience level. Those with a fair amount of * experience know/understand the use of default contexts, however the list has seen many many posts where the implementor is having trouble making things work as expected and a fair number of those have something to do with the proper use of contexts. As with any I/T system, layered security is important including the underlying OS, apps (including *), the network itself, etc. However, there are many systems residing directly on the Internet and none of us have any issues when the systems are properly secured. That is my major concern too, the * config files (as we all know) are not the easiest to read and when the setup becomes more complicated it's difficult to know for sure if you haven't left any loopholes open (for example a caller on hold that can dial outside etc.) Would be nice if there was a script that you could feed an access point to the asterisk server in question (be it SIP or IAX login) and that would just start to try and do anything and report the result). At the same time I realise that this would be a great tool for script kiddies too but I guess they will not be hindered by the lacking of such a script anyways. ___ 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] failed to compile zaptel on redhat
On Sunday 16 January 2005 04:29, Steven Critchfield wrote: > linux/moduleparam.h is actually part of the kernel source. It is created > when you config and compile the kernel. It holds the version symbols > needed to properly link the new drivers into the kernel. No, it is part of the virgin kernel sources and defines the kernel modules parameters api. If he does not have it, then it looks like he does not have the full kernel sources installed. > I suggest you find a kernel compile howto that is at least understanding > of anything specific to the brokenness of RedHat and follow the > suggestions found within. >[...] > > Xu, Duo wrote: > > > why linux/moduleparam.h is missing in the source? I > > > saw it in 2.6 source. B ___ 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] announcing caller id?
> Any one have any solution for this? > We need to have the caller id information announced when the phone is > answered. > for example > I am sitting at my desk, my phone rings. > I pick it up and hear call from 55 to except press 1 to decline The Grandstream BT100 series phones will do this without the help of asterisk. Otherwise, you can write an extension to do it, allowing for calls with callerid disabled of course. ___ 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] NuFone help
> -- Call accepted by 66.225.202.72 (format gsm) > -- Format for call is gsm > -- Hungup 'IAX2/NuFone/1' >== No one is available to answer at this time Is the callerid a number like 7073131 ? ___ 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] announcing caller id?
Any one have any solution for this? We need to have the caller id information announced when the phone is answered. for example I am sitting at my desk, my phone rings. I pick it up and hear call from 55 to except press 1 to decline press to any help would be grately appreciated! ___ 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] How to demo wired phone set on a wireless network
On Sun, 16 Jan 2005, David Norton wrote: > If you know the brand of wireless access point, you are able to try find a > repeater or bridge for that particular brand. Eg. If they using dlink > equipmenk, the DWL2100 can act as a repeater for it. Quite a few brands of > wireless equipment will only bridge to the same brand. > > The second option is to use a "generic" style bridge, ie. One that you are > able to set to act as a "client" on their network. The Senoa client/bridges > are ideal for this and have a good power output too. I believe some linksys > equipment does this too but I have never tried it. > > One thing to be careful of, when you are using a bridge, rather than > client/bridge, some equipment maps its own mac address to the IP address you > use, and will not allow you to have more than one real ip on the other end > of the bridge. This caused a few problems for me Minor nit: you can have as many ip numbers as you want, it is the mac address that is used and limited. Apparently the process of establishing more than one mac address for one wireless client device is not standardized. I know that the Zyxel wireless access points and the cheap zyxel wireless adaptors can work together to connect several mac addresses. One has to read the specifications to find out what works. Peter ___ 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
