Re: [Asterisk-Users] Mini frame before first full voice frame (IAX)
> Has anyone seen this before: > > Jan 24 18:24:56 WARNING[7959] chan_iax2.c: Received mini frame before first > full voice frame http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco phone
> My Cisco 7960g SIP phones share an annoying "feature": Anything I plug into > their 2nd Ethernet or PC port loses connection every thirty seconds or so. I > did a ping -t and can see regular drops. I do not have access to Cisco's > tech support archives. I am pretty sure that this is a simple configuration > problem. Can somebody point me in the right direction? Thanks in advance. It certainly doesn't sound like a feature :) Make sure that the port is configured for a PC and not a hub. I don't have a 7960 to check this on, but I know that there is a way to set the behavior of that port. Methinks it is near the end of the options under "Network Settings." -a ___ --Bandwidth and Colocation sponsored by Easynews.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] free dids on goiax.com
> Unfortunately I had to restrict the free us/canada outbound calling back > down to toll-free only. There was a lot of war dialing and prank > calling going on. I'm working on some stuff to hopefully curb that kind > of stuff down so I can unrestrict outdial again, but this is the problem > with free.. there is always someone that will abuse it. What about unrestricting it for people on the Asterisk list ;) Or, perhaps more useful, what about something similar to a spam filter. Granted, there are many types of and theories for identifying spam and other unwanted messages, but what about adapting something that could be applied to this telephony service? It could learn what sort of "threat" an account poses by examining number of outbound calls in a time frame, duration of said calls, consistency in calling specific numbers, etc. Something that would identify suspicious behaviours/actions... -a ___ --Bandwidth and Colocation sponsored by Easynews.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] Multiple IAX listeners?
> Can multiple soft phones (running on separate computers) be used > simultaneously from the same outside IP address? Yep, should work fine. Consider how any one webserver can handle multiple http requests to port 80. Or consider when what happens when multiple users behind the same nat firewall (it will look to the outside world that they will all be the same public IP) access the same web resources simultaneously...it works fine. -a ___ --Bandwidth and Colocation sponsored by Easynews.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] Hang up call
> For some reason, every morning at 8:30 I get a call on my main extension. > When this call is picked up, it promptly disconnects. Is there some sort of > "Wake up call" or something that may inadvertently be set in * that could be > causing this? It has been happening for quite some time, and I always just > brushed it off, but it's consistency and regularity has caused me to wonder. Asterisk can make calls based on files that appear in /var/spool/asterisk/outgoing You may want to check the contents of this directory around 08:28 or 08:29. If there is a file in there, that's probably why you're getting the call (Asterisk should delete the file after the call is has been answered/timed out). So check the directory around 08:30; if the file is gone then but reappears the next morning, then it sounds like something is causing that file to be generated. If there is never a file in there, then the calls are coming from elsewhere ;) What does callerid show? How about Asterisk console? -Andy ___ --Bandwidth and Colocation sponsored by Easynews.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] WiFi Phones
> Anyone have good words to say about any of the WiFi handsets currently > available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available "security" option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.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] Call-in/Call-out
> How would I setup where I call into my number and
> press say 911 and then it would ask for a pass and
> would accept it and then would prompt for a number so
> I could call out of my number on the road?
How about in whatever context you define your inbound number, add this exten:
exten = NXXNXX,1, Authenticate(911)
exten = NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]:iaxserver.com/1${EXTEN})
exten = NXXNXX,3,Hangup
-a
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Re: [Asterisk-Users] Re: Voice Encryption
> Does Asterisk support encryption of voice traffic? I found following wiki > that describes IAX RSA authentication. I was able to implement the > public/private key authentication among three Asterisk servers connected > using IAX protocol. I am not certain if voice traffic can also be encrypted > among the Asterisk servers. Your help is highly appreciated. There has been a little discussion of this topic on the asterisk-security list somewhat recently. You may want to look at the message archive from August ( http://lists.digium.com/pipermail/asterisk-security/2005-August/thread.html ) as well as the one from September. Hope this offers a little help; as someone mentioned, not much documentation is out there. >From the August thread: >Basically it's an automatic features (CVS-HEAD only, btw) where anytime >a pair of IAX peers have encryption set to "yes" (or, as I understand >it, a mutually-agreed-upon and supported algorithm) then it just happens >automagically. (Brian Capouch) -a ___ --Bandwidth and Colocation sponsored by Easynews.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] Hook Flapping on Cisco 7960
> I've seen this on 4 phones now so I don't think it is hardware related > (like some stupid magnet and reed switch problem). > On the 7910's and on the 7960's, it is a membrane switch. I had a 7960 apart two days ago. -a ___ --Bandwidth and Colocation sponsored by Easynews.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] Hook Flapping on Cisco 7960
Matt: I had this happen to me on a 7910. I ended up taking the phone apart a number of times and finally installed areal spring to try to keep the thing off hook when I was holding the handset... it still doesn't sit perfectly. I think the spring wears down after awhile; if you've ever looked inside, it's just a little piece of spring steel. -Andy On 9/27/05, Matt <[EMAIL PROTECTED]> wrote: > Hey all, > > I'm running * (1.0.9) with some 7960's. When I pick up the handset it > often will flap on and off hook eventually hanging up on me and not > realizing that I am still holding the handset. > > These phones are running with the SIP 7.5.0 image on a LAN (no NAT). > > > I am not sure if this is a phone issue or an Asterisk setting. Ideas? > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: goiax expanded with free us domestic calling
> > Can I ask how you are providing calls to us domestic numbers for free? > > > > goiax.com is backed by TxLink [www.txlink.net]. We terminate a lot of > minutes. Matt: That first logo ( companylogo / www.webaddresshere.com ) on the website could use some work :) but the service works great!! Thanks! -Andy ___ --Bandwidth and Colocation sponsored by Easynews.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: Re: [Asterisk-Users] VoipBuster with astersisk?
On 8/31/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Please ignore my incompetence. what is AMP and where do I read about using it? > > Thanks, > Rudolf > > Rudolf: GIYF: this link should get you started: http://www.google.com/search?hl=en&lr=&q=amp+asterisk&btnG=Search I think there is also a fair amount in the wiki @ http://www.voip-info.org/tiki-index.php -Andy ___ --Bandwidth and Colocation sponsored by Easynews.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] two extensions for same phone
On 7/19/05, Jerry Geis <[EMAIL PROTECTED]> wrote: > All, > > I am intersted in having 2 extensions for the same sip phone... > > The reason is when I am using the outgoing spool file to initiate a > SendText() command > to my SIP phone I dont want voicemail to answer the call and my text > message go there. > > I want my outgoing call to FAIL and then I can try again some time later > to deliver the message. > > Any suggestions on how to do this is appreciated. > > Thanks, > > Jerry > Jerry: It sounds like what you want to do is have the phone register two lines (sip.conf)? Otherwise, you could create two extensions for the phone (extensions.conf). One of them could be for voice calls where you do the whole Dial(SIP/XXX), Voicemail(xxx) Hangup or however it is that you do things on your system. I'm not familiar with the SendText() command, but I imagine if you had a second extension that simply lacked the Voicemail(xxx) part, the message would go through as you desire. -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] SYMBOL NETVISION II NP-3010
On 7/15/05, Barton Fisher <[EMAIL PROTECTED]> wrote: > I was looking at these SYMBOL NETVISION II NP-3010 VoIP TCP/IP WIRELESS > PHONES - I know they have been discontinued. > > Am I asking for trouble to buy some of these for use on Asterisk? > > TIA > > Bart Bart: I purchased some of these a while back for about $30 US and than never got motivated enough, so I can't give any pointers to configuration, except for the actual phone. They seem to be sneaky little devils on the phone for keypad configuration; one of Symbol's cable may be required (it can't be readily made: serial on one end and custom connector that no distributor seems to carry on the other). They are relatively cheap, though. They also only do H.323, so be prepared to play around with that for a bit. -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] No sound
Ronald: If you're using SIP and I'm reading your email correctly, you most likely have some sort of NAT issue. There have been tons of NAT etc. emails on the list; those may help. -Andy On 7/6/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: > I have an asterisk box installed, but all connections to outside of the > private network do not have a sound. > > Can you give me a hint what it could it be? > > > bye > > Ronald > > ___ > 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 sound
On 7/6/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: > I have an asterisk box installed, but all connections to outside of the > private network do not have a sound. > > Can you give me a hint what it could it be? > > bye > > Ronald Ronald: If you're using SIP and I'm reading your email correctly, you most likely have some sort of NAT issue. There have been tons of NAT etc. emails on the list; those may help. -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] Emergency Asterisk Guru Help needed EMERGENCY
Jeffrey: Don't you hate it when that happens?? :) Anyhow, here's what I've got in that directory: [EMAIL PROTECTED] asterisk]# pwd /var/spool/asterisk [EMAIL PROTECTED] asterisk]# du -h 4.0K./voicemail/default/512/INBOX 8.0K./voicemail/default/512 4.0K./voicemail/default/500/INBOX 8.0K./voicemail/default/500 20K ./voicemail/default 24K ./voicemail 4.0K./vm 4.0K./outgoing 4.0K./qcall 4.0K./tmp 1.2M./monitor 1.2M. [EMAIL PROTECTED] asterisk]# -Andy On 7/6/05, Jeffrey Starin <[EMAIL PROTECTED]> wrote: > 911 Help! > > I accidentially deleted all directories under /var/spool/asterisk > > I did use the backup facility not too long ago but cannot find the > process for restore. > > However, I don't believe a full restore is needed -- I just need to know > the names of the directories under /var/spool/asterisk and re-create > them (I hope!). Can some kind soul give me some direction or tell me > the directory structure under /var/spool/asterisk? > > Thanks, > > 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
Re: [Asterisk-Users] Need help connecting two * pcs with *@home
On 6/13/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> What I want to be able to do is dial ext 2009 from server b (2009 doesnt exist
> on this server) and have it ring 2009 on server a. If that is difficult to do,
> I can always revert to dialing a prefix, eg. 22009 to call server a.
If you're using IAX, you may find the following of some use:
Server A:
In iax.conf:
[general]
register => andy:[EMAIL PROTECTED]
[andy]
type=friend
context=otherservervoip
auth=rsa
inkeys=otherserver
In extensions.conf:
[otherservervoip]
exten = _66XXX,1,Dial(IAX2/andy:[EMAIL PROTECTED]/${EXTEN:2})
Server B:
[andy]
type=friend
secret=mypassword
username=andy
host=dynamic
context=default
outkey= otherservervoip
I gleaned most of this from the wiki (voip-info.org). Highly
recommended and valuable resource. Be prepared to play around with
this stuff a bit first before it works. I think you'll find the
documentation extremely helpful.
-Andy
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Re: [Asterisk-Users] Asterisk to Cisco Voip System Unity
Simone: You should be able to to this fairly easily using h.323 extensions between CCM and Asterisk. Check the list archives; I think other people have had good luck with this. -Andy On 6/9/05, Simone <[EMAIL PROTECTED]> wrote: > Hi all, first post. My company's office in the UK is soon going to get a > Cisco VoIP solution system. What I am interested in, and couldn't find > googling, is if it is possible to connect an Asterisk solution to the > Cisco system and have all the nice advantages of it (mainly calling the > extensions and directly reach the other office). > > Thanks, have a nice day > > Simone > ___ > 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] DID Routing over SIP
I imagine that if your Asteirsk box registers all 10 of the DIDs, then when a call comes in on one of them, you could tell Asterisk to dial the location where you want that call to be routed. I think CCM/CCE only support SCCP, MGCP, and H323, however. Unless you were planning to have an Asterisk box as a SIP <-> MGCP device to interface with CCE. -Andy On 6/2/05, Asterisk VoIP <[EMAIL PROTECTED]> wrote: > Hi- > > Does any one know how to route DIDs over SIP from one Asterisk box to > other SIP gateway. I have no Zap channel or PRI on my asterisk box. I > got 10 DIDs from one VoIP provider and would like to pass ( > route/forward) 2 each to my friends SIP gateway ( Asterisk/Cisco > Express call manager). > > I am new to Asterisk. I would really appreciate if some one help me to > get this through. > > Thanks, > Mike. > ___ > 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] Incoming and Outgoing
Bill:
Try adding this extension in whichever context you handle your inbound calls:
exten = _1NXXNXX,1,Authenticate(1234)
exten = _1NXXNXX,2,Playback(pls-wait-connect-call)
exten = _1NXXNXX,3,SetCallerID(${CALLERIDNUM}); Set your CallerID
as a ten digit $
exten = _1NXXNXX,4,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com NANPA
This example requires a password (1234) in order to make a call. This
prevents people from calling in and then making outbound calls on your
accout. This example also assumes you use voipjet (which I have found
to be quite a solid termination provider). If your provider doesn't
allow you to set your outbound caller ID, you will want to remove that
line.
Hope this helps,
Andy
On 6/1/05, Bill Madeira <[EMAIL PROTECTED]> wrote:
> I''m trying to setup asterisk to answer the call and when recording is
> played, i am able to dial a number of some sort and I will be able to use
> that line to dial out. My cell phone's long distance is not part of my plan
> so i would like to take advantage of my home asterisk to be able to do this.
>
> bill
>
>
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Re: [Asterisk-Users] Suppress "Missed Calls" 7960 SIP
Rob: Not sure how to (though I agree it would be handy). If anything, it would be a Cisco thing. Have you checked their website to see if the have any tips? -Andy On 5/31/05, Robert Goodyear <[EMAIL PROTECTED]> wrote: > Does anyone know how to suppress the "Missed Calls" indication -- > perhaps on a per-line basis -- on the 7960 running SIP? > > Reason: I've configured a group of extensions to ring for inbound calls > and it seems pointless to accrue missed calls on those line > presentations. > > /rg > > ___ > 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] Cisco 7960s and skinny
Yusuf:
This should be a functioning /etc/asterisk/sccp.conf :
==
[general]
keepalive = 70
context = default
dateFormat = Y-M-D ; M-D-Y in any order (5 chars max)
bindaddr = 192.168.1.167; replace 1.2.3.4 with the ip address of the
asterisk box.
port = 2000 ; listen on port 2000 (Skinny, default)
[SEP000123456789]
; This device is a 7910, a single line model
; from cisco.
type= 7910
autologin = 524
description = 7910
[524]
id = 524
pin = 524
label = South Lobby
==
Here is my /tftpboot/XMLDefault.cnf.xml file.
I'm not sure if the 7910 actually gets any useful information from it
(at least in the case of Asterisk as the CM). I do not have a
SEP000123456789.cnf file or anything like that: I did at one point and
it would cause the phone to not boot, if I recall correctly.
==
2000
192.168.1.167
http://192.168.1.167/asterisk/ciscomenu.xml
P00403020214
P00303020214
P00303020214
==
And, of course, here is a snippet from my /etc/asterisk/extensions.conf:
==
[default]
exten = 524,1,SetCalledParty("South Lobby" <${EXTEN}>)
exten = 524,2,Dial(SCCP/524,30,Ttr) ; You may want to change some of
these options
exten = 524,3,Hangup
==
Hope that helps. Lemme know if you need anything else.
-Andy
On 5/21/05, Yusuf Iqbal <[EMAIL PROTECTED]> wrote:
> Hi Andy,
> I have been trying to get 7910's work with *. I have tried with both
> skinny and chan_sccp. Could you please instruct me about the
> configuration? I have found some detalis about 7920 with sccp in
> voip-info.org. But I haven't find any document for 7910. Please help
> me to get them work properly. I have four 7960's and working fine with
> SIP.But I am in trouble with those (total of 8) 7910 IP phones.
>
> On 5/2/05, Andy Hamilton <[EMAIL PROTECTED]> wrote:
> > Anton:
> >
> > Yes, the whole legally licensing the phones from Cisco is a major drawback.
> > I just compiled and install the chan_sccp-mayday CVS snapshot.
> > It works like a charm (thanks Julien!). In fact, I did some dialing
> > and redialing to my 7960 (SIP) from my 7910 and the 7960 ended up
> > needing to be rebooted. I was quite pleasantly surprised; the
> > 7910/chan_sccp seemed to be having a field day.
> >
> > Here is what I have found for the 7910:
> > -> Hold button works (a little weird when more than one call)
> > -> Line button brings up the line
> > -> Transfer button doesn't seem to work
> > -> Msgs button doesn't seem to work
> > -> Conf button doesn't seem to to work
> > -> Forward button doesn't seem to work
> > -> Speed1 doesn't seem to work
> > -> Speed2 doesn't seem to work
> > -> Redial works
> >
> > Hope this helps.
> >
> > -Andy
> >
> > On 5/1/05, Julien Goodwin <[EMAIL PROTECTED]> wrote:
> > > On Sat, Apr 30, 2005 at 01:00:18PM -0500, Andy Hamilton arranged a set of
> > > bits into the following:
> > > > I'll be able to get back to you Sunday night about specifics; the
> > > > phone is not where I am right now. Using chan_sccp, (I think November
> > > > 2004 or so CVS Head) I know I can receive calls, place calls, etc. It
> > > > is a rather low volume phone, so I don't know off hand about specific
> > > > keys; I'll check those later.
> > > Generally if the phone supports the function, and support is in
> > > chan_sccp for that function it will work for all phones.
> > >
> > > > Additionally, I have not yet tried a new copy from CVS.
> > >
> > > > Occasionally, I think the chan_sccp driver blips out in Asterisk (it
> > > > may be the phone; I've had it apart several times because the on/off
> > > > hook switch membrane is a little sketchy). I have dealt with this by
> > > That's one of the big things that causes problems, both with chan_sccp
> > > and the phones themselves, both get a little confused. However several
> > > other crash issues have been recently fixed, so running CVS_HEAD is
> > > advised.
> > >
> > > > restarting Asterisk. The only other thing I can say right now about
> > > > the 7910 is that it and my Cisco FastHub don't get along. At all. I
> > > > have the 7910 plugged into my 7960.
> > > That's odd, the only time I've ever had ethernet incompatabilities was
> > > with a very cheap switch.
> > >
> > > > Overall, I would say that if you have a non-critical system and would
> > > > lik
Re: [Asterisk-Users] Voice Quality
David: Bandwidth may be an issue; however, do you have any timing devices installed? Digium's hardware (or any generic knockoffs) will provide this. There are also some other ways, such as ztdummy or a usb controller (haven't used either of these, so I don't know any specifics. Check the Wiki). -Andy On 5/3/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hello, > > I have setup two * servers and they are communicating using IAX. I'm > passing calls from SRV A (internet connection T1) to SRV B (internet > connection: 512). > > For some reasons I have an issue with the quality. The voice is a bit > scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. > > Now, assuming that I have an issue with Bandwidth, what would be the best > way to configure my iax.conf. (A bit confused about jitterbuffer and tos) > > Here is my iax.conf @ location A: > > [general] > port=4569 > bandwidth=low > disallow=all > allow=ilbc > ;allow=ulaw > ;allow=speex > jitterbuffer=200 > jitterbuffer=yes > tos=lowdelay > > and iax.conf @ location B: > > [general] > port=4569 > bandwidth=low > disallow=all > allow=ilbc > ;allow=ulaw > ;allow=speex > jitterbuffer=200 > jitterbuffer=yes > tos=lowdelay > > [guest] > type=user > context=default > callerid="Guest IAX User" > disallow=all > allow=ilbc > > Thanks guys > > ___ > 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] Light weight and slimmed Asterisk
Kumara: Knock it out in your modules.conf noload => chan_modem.so You can use this for any modules. Just be careful, Asterisk may rely on some that you don't think you need. Check these pages on the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20modules.conf http://www.voip-info.org/wiki-Asterisk+modules -Andy On 5/3/05, Kumara Jayaweera <[EMAIL PROTECTED]> wrote: > Greetings to all! > Sorry for the numerous postings. but How could I slim my Asterisk PBX. > Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't > have any special hardware. Please, could I hope your various suggetions in > this regards. brief me your idea. > > ___ > 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] email notification when leaving a message
Alex: Depending on your server's IP address/DNS and the destination email address, it's possible that the remote server has blacklisted mail originating from your IP address. I think this is most notable when email is being sent from Asterisk servers on residential connections, cable/dialup/whatever. You may be receiving failed sending notifications in your root mailbox on your Asterisk server; this would confirm what is happening. -Andy On 5/3/05, Alexandre Charles <[EMAIL PROTECTED]> wrote: > Hi! > > I have configured: > iax.conf; > voicemail.conf > extensions.conf > > everything works fine... the only things.. > > i do not receive any email notification when a > voicemail is left on the *.. any clues??? i think my > email server works(?).. In fact i am able to send an > email to the root (mail root etc...).. but aside > that.. i am not able to send any other email outside > the * box... any clues on how to solve that... > I have installed * on RedHat.. > > Thanks for your help, > > AlexC > > __ > Lèche-vitrine ou lèche-écran ? > magasinage.yahoo.ca > ___ > 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] Cisco 7960s and skinny
Anton: Yes, the whole legally licensing the phones from Cisco is a major drawback. I just compiled and install the chan_sccp-mayday CVS snapshot. It works like a charm (thanks Julien!). In fact, I did some dialing and redialing to my 7960 (SIP) from my 7910 and the 7960 ended up needing to be rebooted. I was quite pleasantly surprised; the 7910/chan_sccp seemed to be having a field day. Here is what I have found for the 7910: -> Hold button works (a little weird when more than one call) -> Line button brings up the line -> Transfer button doesn't seem to work -> Msgs button doesn't seem to work -> Conf button doesn't seem to to work -> Forward button doesn't seem to work -> Speed1 doesn't seem to work -> Speed2 doesn't seem to work -> Redial works Hope this helps. -Andy On 5/1/05, Julien Goodwin <[EMAIL PROTECTED]> wrote: > On Sat, Apr 30, 2005 at 01:00:18PM -0500, Andy Hamilton arranged a set of > bits into the following: > > I'll be able to get back to you Sunday night about specifics; the > > phone is not where I am right now. Using chan_sccp, (I think November > > 2004 or so CVS Head) I know I can receive calls, place calls, etc. It > > is a rather low volume phone, so I don't know off hand about specific > > keys; I'll check those later. > Generally if the phone supports the function, and support is in > chan_sccp for that function it will work for all phones. > > > Additionally, I have not yet tried a new copy from CVS. > > > Occasionally, I think the chan_sccp driver blips out in Asterisk (it > > may be the phone; I've had it apart several times because the on/off > > hook switch membrane is a little sketchy). I have dealt with this by > That's one of the big things that causes problems, both with chan_sccp > and the phones themselves, both get a little confused. However several > other crash issues have been recently fixed, so running CVS_HEAD is > advised. > > > restarting Asterisk. The only other thing I can say right now about > > the 7910 is that it and my Cisco FastHub don't get along. At all. I > > have the 7910 plugged into my 7960. > That's odd, the only time I've ever had ethernet incompatabilities was > with a very cheap switch. > > > Overall, I would say that if you have a non-critical system and would > > like to use a 7910, chan_sccp should be able to handle it fine. > > However, if you budget permits, the 7960 and 7940 phones are quite > > nice (use SIP with those -- it's far more reliable. I must say, > > though, that my 7960 has frozen/crashed a handful of time when running > > the SIP image. That was the phone itself, Asterisk was fine.) I have > > yet to purchase a 7905 or 7912, but I've played around with some > > 7912's on a CCM system -- they seem quite nice and I think they take > > SIP. > Yep, they do. (Don't know about the 7902, but really can't see why > anyone would buy one) > > >The 7920 is also nice because it's wireless. However, I don't > > think Cisco has anything but a Skinny image for it [yet]. > No they don't, and forget the yet, if a phone isn't announced with SIP > support it probably never will have it (witness: 7935/6, 7970) > > > I would stick with SIP wherever you can. > And I agree > > Thanks, > Julien > chan_sccp project lead > > > ___ > 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] Cisco 7960s and skinny
Anton: I'll be able to get back to you Sunday night about specifics; the phone is not where I am right now. Using chan_sccp, (I think November 2004 or so CVS Head) I know I can receive calls, place calls, etc. It is a rather low volume phone, so I don't know off hand about specific keys; I'll check those later. Additionally, I have not yet tried a new copy from CVS. Occasionally, I think the chan_sccp driver blips out in Asterisk (it may be the phone; I've had it apart several times because the on/off hook switch membrane is a little sketchy). I have dealt with this by restarting Asterisk. The only other thing I can say right now about the 7910 is that it and my Cisco FastHub don't get along. At all. I have the 7910 plugged into my 7960. Overall, I would say that if you have a non-critical system and would like to use a 7910, chan_sccp should be able to handle it fine. However, if you budget permits, the 7960 and 7940 phones are quite nice (use SIP with those -- it's far more reliable. I must say, though, that my 7960 has frozen/crashed a handful of time when running the SIP image. That was the phone itself, Asterisk was fine.) I have yet to purchase a 7905 or 7912, but I've played around with some 7912's on a CCM system -- they seem quite nice and I think they take SIP. The 7920 is also nice because it's wireless. However, I don't think Cisco has anything but a Skinny image for it [yet]. I would stick with SIP wherever you can. -Andy On 4/30/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Andy > > How did the 7910 worked with skinny under *? Did all the keys on the phone > worked? Ive seen sometimes the forward key or something does not fully do > what you would excpect. > > What are the drawbacks from using skinny vs sip under *? > ___ 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] Console Warning Message
http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.html On 4/28/05, Daniel Salama <[EMAIL PROTECTED]> wrote: > Does anyone know what this mean? > > Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received > mini frame before first full voice frame > > Thanks, > Daniel > > ___ > 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] Checking for a sound file
Marc: You may be able to do this with voicemail; you'd get the authentication, realtime outbound message alteration, etc. You would just have the DID support number go straight to voicemail. The only think left would be to see if there's a way to not permit a message to be left. -Andy On 4/26/05, Marc <[EMAIL PROTECTED]> wrote: > > Hi, > > At this moment I'm configuring my first asterisk pbx, and am running into > the following "problem": I would like to create a phonenumber for my > customers, which they can call to hear if there are any problems with the > servers. In case of a problem, I would like to be able to call that number, > authenticate myself and record a new message. From that moment that message > must be played when customers call. When the problem is solved, I would like > to call the same number again, authenticate, and remove that message, so the > original message is again played to customers that call. > > I've read the wiki pages, but I'm not able to create this configuration. Can > somebody please give me some tips how to do this? > > Regards, > Marc > ___ > 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] Basic Setup Question
Terry: You'll probably want to set up an IAX2 channel between them. Here is a good place to start looking: http://www.voip-info.org/wiki-Asterisk+IAX+Channels Also, I think this list has had a fair share of IAX2 inquiries recently. If you check the archives (http://lists.digium.com/pipermail/asterisk-users/), I think you'd find something useful. If you still have problems once you have tried Wiki and list suggestions, feel free to send a message to the list with specifics. Hope that helps, Andy On 4/21/05, Terry Bomersbach <[EMAIL PROTECTED]> wrote: > I have two Asterisk boxes installed but am not sure how to setup the > configuration for what I want to do. > > One box has two FXO cards in it that will connect two PSTN lines. I want to > have Asterisk transfer the incoming calls to the other box which has an FXS > card in it. That box should ring that call through to a handset attached to > the FXS card. > > Any advice on how to make this work would be 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 > ___ 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] Cisco 7920 - chan_sccp - asterisk@home .9
Chuck: I have been able to use a 7920 with Asterisk. Never used [EMAIL PROTECTED] If you post your config files (sccp.conf, SEPX.cnf, etc), I can have a look at them for any suggestions. -Andy On 4/19/05, Chuck Smith <[EMAIL PROTECTED]> wrote: > Has anyone been able to get chan_sccp to work with a 7920 on [EMAIL PROTECTED] > > I was able to compile chan_sccp and I see it as a running module in Asterisk > but I don't see any debugs on the console that shows the phone even making > an attempt. I followed the examples to the T but still no luck. > > Thanks > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell > Sent: Tuesday, April 19, 2005 10:39 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] US$200 bounty for * paging feature > > On 4/19/05, Mike <[EMAIL PROTECTED]> wrote: > > >> . close source and we own the code. > > You are no better then Microsoft. > > Speaking of an over reaction > > -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 > > ___ > 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] Upgraded now Asterisk won't start
What it you try to downgrade, i.e. go back down to version 1.0.6? -Andy On 4/18/05, Paul A Brown <[EMAIL PROTECTED]> wrote: > I have just installed 1.0.7 from the debian source using apt-get. > > Now it won't run. I try to start it, do a PS and its there for a min then > disappears. asterisk -r says can't connect to remote server and asterisk -c > says > > Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k, Copyright (C) 1999-2004 Digium. > Written by Mark Spencer <[EMAIL PROTECTED]> > = > [ Booting.res_odbc loaded. > .Registered Config Engine odbc > res_config_odbc loaded. > .No agent configuration found -- agent support disabled > .Unable to load config mgcp.conf, MGCP disabled > .Unable to open IAX timing interface: No such file or directory > No IAX provisioning configuration found, IAX provisioning disabled. > ..Unable to load config skinny.conf, Skinny disabled > .Unable to load config phone.conf > chan_phone.so: load_module failed, returning -1 > Loading module chan_phone.so failed! > > I can't find any errors in /var/log/asterisk (Not sure which file should > show errors but none do anyway) > > I am bound to of done something daft and apologise in advance. > > If anyone can help I would appreciate it > > Paul > > ___ > 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] Cisco 7940
Thomas: It sounds like you may need to unlock your phone. If I recall, you can hit **# to unlock it; then go to the settings menu. On newer firmwares, you'll have fun trying to get past an actual password. Check http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx as well as this list's archives from April 14,15,16,17 of this year. This topic just came up recently a few times. -Andy On 4/18/05, Thomas RULMONT <[EMAIL PROTECTED]> wrote: > Hello list, > > Could you tell me if you ever succeeded in configuring Cisco 7940 and > chan_skinny. How ? (I cannot configure my phone, almost any submenu is > unavailable) > > Thx. > > -- > > > Thomas RULMONT > Responsable Commercial > Alterys SA > > T. +32 87 325939 > T. +32 486 863216 > E. [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 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] VOIP to PTSN provider
Very true. I have found the outgoing CID to be very ... useful. Although occasionally inconsistent on the remote party's end, even though voipjet's CDR shows the CID string that I sent. -Andy On 4/17/05, Gregory Wiktor - ADCom Corp. <[EMAIL PROTECTED]> wrote: > I have to agree that voipjet is a good service. If only they had did's > it would be even better, but I like the fact that outgoing cid works > well. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ed > Greenberg > Sent: Saturday, April 16, 2005 6:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] VOIP to PTSN provider > > 800 numbers are free to the caller because the recipient pays the > charge. > > Voipjet has no way to get paid anything for carrying the calls, hence > they are unwilling to use their resources to move calls with no > revenue. > > Can you blame them? :) > > > > --On Saturday, April 16, 2005 9:44 PM +0100 Chris Hills > <[EMAIL PROTECTED]> wrote: > > > Andy Hamilton wrote: > > > >> I use voipjet and am quite pleased. Good enough rates and no > >> noticeable quality issues. > >> > >> http://www.voipjet.com > >> > >> Plus, you can even test it before you buy. > >> > >> > > On their pricing page, they have:- > > > > There are some providers who can terminate some, but not all, 1800 > > numbers for free. (If they could terminate all 1800 numbers for free, > > then we'd use them!) > > > > I don't understand - I thought all 1800 numbers were free? > > ___ > > 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 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] VOIP to PTSN provider
No, I suppose you can't blame them What is FWD's motivation (or IAXtel, etc) to provide this service, then? -Andy On 4/16/05, Ed Greenberg <[EMAIL PROTECTED]> wrote: > 800 numbers are free to the caller because the recipient pays the charge. > > Voipjet has no way to get paid anything for carrying the calls, hence they > are unwilling to use their resources to move calls with no revenue. > > Can you blame them? :) > > ___ 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] large analog to asterisk
The cat 3 issue depends on your phone, if you went with VoIP phones, you would need to make sure that it could be set/forced to 10Mbps. I have only used Cisco phones, and, save the 7910 and 7902 (may a few others), they all are fully capable of doing 100Mbps because of their internal switch (you can colocate a comptuer with the phone); you'd need to tell them to go 10. Cisco hardware would also be a huge investment for the hotel, most likely way more than what the average guest needs. Also, just FYI, the 7971does... gigabit ethernet!! That's pretty sweet. -Andy PS - Is the currently installed wiring cat3? On 4/16/05, Michael D Schelin <[EMAIL PROTECTED]> wrote: > Good point. Here is another Suggestion. Why not use the existing analog > phones to their PBX and go out to channel banks for their phone line > trunks. Then go to Asterisk for the rest. They don't have 700 trunks. > This will save on equipment costs and you will get some of the benefits > of Asterisk. It's not the best solution but it will work. Also if you > do want to replace all you system with new phones then try my idea of > using the cat 3 cable. As far as the switches gos, remember it's not the > cable the determines the speed but the equipment connected to it. I > have yet to see a sip phone above 10 Mb. So you can disregard Mr. > Hamilton's statement about the switch. Yes you investment will be high, > but that is a business expense. I'm currently doing that cat 3 trick. > Don't worry about your customers connecting to your phone system. They > won't know it's IP. > > > John Novack wrote: > > > ___ 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] VOIP to PTSN provider
I use voipjet and am quite pleased. Good enough rates and no noticeable quality issues. http://www.voipjet.com Plus, you can even test it before you buy. -Andy On 4/16/05, Ilija Poznic <[EMAIL PROTECTED]> wrote: > I am trying for 1 month to use IConnect with * an I am not satisfied with the > sound quality. Can someone to recommend me some VOIP to PTSN provider. > > Thanks a lot Ilija Poznic > > ___ > 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] How do I connect my Asterisk PBX to a service Provider
This depends 100% on what kind of service provder you have. If you are going with a VoIP provider that mentions Asterisk, it is entirely possible that they will tell you what you need to add and to which files. In regards to your numerous posts, may I kindly recommend the Wiki? It should tell you everything you need to know about adding extensions, configuring clients, and potentially interfacing with a service provider. Check: http://www.voip-info.org/tiki-index.php?page=Asterisk -Andy On 4/15/05, Kumara Jayaweera <[EMAIL PROTECTED]> wrote: > Hi List, > How do I connect my Asterisk PBX to a service Provider? suppose I have a > service provider. (which file should I edit for this in my box) > > Thanks > > Kumara > > ___ > 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] large analog to asterisk
And then you'd need to purchase 700 VoIP phones; not a small investment. With all due respect to Mr. Schelin, I think the analog method may be best, unless you plan to expand the services that you offer to the guests. If the rooms did have cat3, you could eventually expand your offering to include internet access for the guests, advanced phone features (on the IP phones), etc... I stray from the topic. You'll be facing some sort of hardware investment aside from the server, I think, and that is either IP Phones or a lot of hardware to support the analog lines, per Rusty's suggestion. Granted, this will be a large project, I think it would be wise to weigh the benefits of going to IP Phones now that will most likely go mainstream in the near future or support an aging but solid analog technology. -Andy On 4/15/05, Michael D Schelin <[EMAIL PROTECTED]> wrote: > If your rooms analog phones are wired with cat 3 cabling you can do 10 > Mb over it. Convert all the rooms to Ethernet and use large switches. > One Asterisk box should do the trick. Remember not every room will be > using the phone system at the same time. This should work for you. > > > shane fowler wrote: > > > we are looking at the ability of being able to convert large phone > > system over to asterisk or if it's possible at all. The building is > > two sections containing a large office section (with data cabling) and > > the second section is a hotel with no data cabling. The first section > > is a no brainer with sip hard and soft phones but the hotel part is > > where the problem lies. > > > > The current count of rooms in the hotel is about 600...that's at a > > minimum 600 analog connections. Some rooms have 2-3 phones so as a > > rough number i'm saying 700 total. I see where some people use the > > Adit 600 to do up to 48 analog connections that trunks over 2 T1 > > connections back to asterisk but for 700 phones thats 15 Adits with 30 > > T1'show in the world would you do that?? just several asterisk > > servers with 2-3 Adits per server? is there any other way? I'm open > > to suggestions. > > > > Thanks.. > > > > Shane > > > > > > ___ > > 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] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail
Right here. I find this quite upsetting. You might think that after telling Gmail that a hundred messages are not spam, let alone having a filter to apply a label to all messages with [Asterisk-Users], would be enough. Apparently not. I'm also not aware (correct me if I'm wrong) of any method to tame the spam filter for my particular mailbox, but although Gmail is grand, the spam filter sure doesn't seem to be a learning one. It would also appear that Asterisk-Bis, Asterisk-BSD, and Asterisk-Security are unaffected. -Andy On 4/15/05, Sig Lange <[EMAIL PROTECTED]> wrote: > > Starting around Apr 14th Gmail has started marking all messages for > Asterisk-Users as spam. Prior to that on google groups someone created a > asterisk-test group (seperate from this group). Is this perhaps related? I > believe it all has happened within a week time frame. Gmail is a great > service but if this is what's going to happen I will quit using gmail. I'm > giving a shot out to see any other gmail users out there having this > problem. My Asterisk-Dev seems to be unaffected. > > Who's having similiar issues? > > TIA, > Sig > > ___ > 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] cisco 7960 SIP setup
Mike: I know this sounds patronizing, but do you have the SIP image files? If so, what version? Per the Asterisk wiki page on the 7960/7940s, you may need to upgrade incrementally. Additionally, make sure you have the correct files in the root directory of your tftp server (for linux, this is probably /tftpboot). Also make sure that the tftp server works (you can test it from a linux client). Check the wiki out at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx -Andy On 4/14/05, mk111 <[EMAIL PROTECTED]> wrote: > I can't get the 7960 to reconfigure and work. I am a newbie to voip. I > went through the list and read some other comments about the 7960 and > unlocking it. It is a used 7960 that came with CallManager. I need to > have SIP. I first reset the phone to factory defaults then I changed > the TFTP server address in the settings. I have unlocked the phone with > **# and it shows the lock as unlocked in the upper right hand corner. I > was told that the phone should be able to download the SIP... file once > the TFTP address was changed. So far nothing though. Any ideas? > > Mike > > ___ > 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] Asterisk@home first experience
Bruno: Are you getting any errors or warnings at the CLI? -Andy On 4/14/05, Bruno Quintas <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi all, i installed [EMAIL PROTECTED] v0.8, very clean install (great > piece of software!). > I have successfully configured incoming calls, but i still have some > problems. > My setup is a X100 connected to a POTS, one analogic phone, 2 SIP's > (X-Lite) in Windows, and one IAX in Linux. > > I can make outgoing calls from the hardphone with no problems, but i > can't make them from the softphones, at this time i only want to call > to the PSTN via the SIP's and the IAX, some problem with dialing > rules?? Altough i can get external calls (from PSTN), in the SIP's. > > Can somebody give me some hints? > > TIA > > Bruno > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.2.4 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org > > iD8DBQFCXlxoWn7D5oqv6YoRAstIAJ4zkjeYcVXxf9laI3RGlxidf3KyZwCfb6nt > pMqFP6MBe9M+lRpzFiupX3Q= > =fg+C > -END PGP SIGNATURE- > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ 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] delay problem in asterisk
You may want to see if your provider has other servers you can connect to that have lower ping times or chagne providers. I think 250ms is generally considered the max tolerable delay in voice conversations; you want a ping time substantially less than this. -Andy On 4/14/05, wassim darwish <[EMAIL PROTECTED]> wrote: > i have asterisk on my system and when making a call a > delay problem in talking appears,that means when i > talk to somebody he will listen me after almost a > second (the ping on my voip provider's IP is 700ms to > 800ms)so i dont know if the problem is in the nternet > connection or another problem ,please help me i dont > know what to do. > > __ > Do you Yahoo!? > Yahoo! Small Business - Try our new resources site! > http://smallbusiness.yahoo.com/resources/ > ___ > 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] Cisco 7960s and skinny
Simon: I have had Skinny going on a 7960 (which I then reimaged to SIP). I currently run a 7910 on Skinny (using chan_sccp) and use the aforementioned 7960 simultaneously. Since you mentioned that you will have 50 phones, I assume you are using them in a business setting. I would *highly* recommend using SIP, as I have found that the skinny driver is not as reliable as it could be (not criticizing Jan or Julien at all, here). Reimaging the 50 of them should only take a while (depending on what version of CCM they have at the moment). I reimaged 12 phones once for a business and it took less than 30 minutes after I got it going (toying with the phones to get them to take the image, exactly how the config files were to be set up, etc...). I imagine you could easily get the whole thing done in less than a day (reimaging and config files), then figure out your dialplan. Then there is the whole issue of writing the config files...but you'd have to do those with Skinny, anyhow. I think with SIP you'll have much better reliability. -Andy FWD: 428725 On Apr 12, 2005 12:48 PM, Morris, Simon <[EMAIL PROTECTED]> wrote: > > > Hello, > > Does anyone else have * running with Cisco 7960 phones and skinny? > > All the advise I am reading so far is telling me to load the SIP image on > the phone but I'd like to know what I'm going to lose by persisting with > skinny > > (Not reimaging 50 phones is one benefit amongst others of skinny) > > Thanks for any comparisons you can provide > > Rgds > > ~sm > ___ > 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] Getting CVS HEAD
Here's an excerpt from that page. Obviously, the hyperlinks are missing for some things, but I would reccommend rereading the page, specifically where it says, "...download a tarball of the released sources..." These are release versions. If you want the CVS Head version, perhaps where it says, "To check out code from our CVS repository:" would be the place to look. Download Asterisk You can download a tarball of the released sources at ftp://ftp.asterisk.org/pub/asterisk. You can download the tarball files directly here: Asterisk Zaptel Libpri Asterisk-addons Asterisk-sounds You'll need Asterisk, and if you're using Digium's hardware you'll need zaptel. For T1 or E1 interfaces you'll also need libpri. You will need bison in order to build Asterisk. The ncurses and ncurses-devel packages are required if you wish to build the new tools (e.g. astman). Installation should be in this order: zaptel, libpri, Asterisk The fastest way to obtain Asterisk is to use CVSup. To check out Asterisk using CVSup, create a sup file as follows: *default host=cvs.digium.com *default base=/usr/src *default release=cvs tag=. *default delete use-rel-suffix asterisk libpri zaptel Perhaps call it "asterisk-sup" and put it in /usr/src Then simply: # cd /usr/src # cvsup asterisk-sup Or, you can obtain Asterisk by checking out a fresh copy from our CVS Server. To check out code from our CVS repository: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk On Apr 11, 2005 10:27 AM, Guillermo Salas M <[EMAIL PROTECTED]> wrote: > Hi, I want to download the CVS HEAD version. Any one can show how to get > this version ? > > Is the version from: http://www.asterisk.org/index.php?menu=download the > CVS Head version? > > How I can check if my version is CVS HEAD or not? > > Best Regards, > > -- > Guillermo Salas M. > Telconet S.A. Manta > Calle 15 y Av. 24 Esq. > Phone : 593 5 262 8071 > Mobile: 593 9 985 5138 > e-mail: [EMAIL PROTECTED] > www : http://www.telconet.net > http://www.telcocarrier.net > > Linux User: 255902 > Soporte en Linea en http://www.manta.telconet.net > > Please avoid sending me Word or PowerPoint attachments. > See http://www.fsf.org/philosophy/no-word-attachments.html > > ___ > 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] ignorepat changing the sound of dialtone
This depends on what kind of phone you are using. With most (any?) SIP phones, nothing will be sent by the phone to the server until it actually dials (whereas Skinny phones sent out on/off hook and digits realtime). If you're using a Cisco phone with a sip image, my guess is that you can set something in dialplan.xml (or whichever file it is that the Ciscos look at to match numbers). That way, if you wanted someone to press 0 to get an outside line, the phone would see that an 0 was pressed and immediately dial. Obviously, you would catch this in Asterisk. Once *'s gotten the "call" from an extension that has dialed an 0, an RTP stream with the phone would commence and * would wait for the number that the party wishes to call (I believe there is a setting to change the tone... check your conf files or the wiki; I'm not sure where exactly it is. Once * matches a dialing pattern, it would dial out. Hope this helps. Andy Hamilton FWD 428726 On Apr 10, 2005 7:30 AM, Thomas Andrews <[EMAIL PROTECTED]> wrote: > Howdie folks, > > Is it possible to play a different dialtone as soon as a user dials say > '0' for an outside line ? Ignorepat is an inadequate solution because > local users are accustomed to getting a specific PSTN dialtone. I need > an audible change in the frequency/modulation of the tone. > > Thanks, > Thomas > ___ > 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] Call from publicIP to PrivateIP
I imagine that you are using SIP, which has numerous issures with NAT. Consider using IAX2; one of it's benefits is working with NAT, which I gather is your problem. -Andy FWD:428725 On Apr 8, 2005 7:06 AM, Kamran Ahmad <[EMAIL PROTECTED]> wrote: > hello > > Any one know how to resolve NAT issue. > > PublicIp(UA)->Asterisk on > publicIP-->privateIP(UA) its not working > > PrivateIP(UA)->Asterisk on > publicIP-->publicIP(UA) its working > > how to reslove this issue > > Thanks > Kamran > > __ > Do you Yahoo!? > Yahoo! Personals - Better first dates. More second dates. > http://personals.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] How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?
Ronald: Try reducing the duration for which the wakeup call waits on the person it is calling; my default is 20 seconds. Here is a sample of one of my wakeup call files, automatically generated: Channel: SIP/500 MaxRetries: 1 RetryTime: 10 WaitTime: 20 Application: MusicOnHold Callerid: * Wakeup Call Twenty seconds works for me (without the call going to VM), but I tried setting it to 200. Asterisk said is was an invalid time and the call timed out and hungup after about 40sec. It also looks to me, on the test I just ran, that it will try calling twice, even though MaxRetries is set to 1. -Andy On Apr 6, 2005 5:30 PM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: > We use wakeup calls for reminders, but it happens, that the person to be > reminded is on the phone. To get a voicemail later is not really useful > anymore, ... > Is there a way to avoid that? > > bye > > Ronald > > ___ > 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] Cisco 7940 Outgoing Audio
Which skinny driver are you using? Also, what about incoming calls? Post your skinny.conf or sccp.conf as well as the config file for your 7940 from the TFTP server. It may help also if you include any other generic Cisco config files from your TFTP directory, for example an OS79XX.TXT. -Andy On Apr 6, 2005 3:27 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote: > > > > I'm a Cisco 7940 phone using SCCP. My setup is a private network with the * > box acting as dhcp server and also tftp server. The phone loads and dials > out fine. I can hear the other person, but there is no outgoing audio. > I've read that this is an RTP problem and have tried making some changes in > /etc/hosts to point to my * box IP but with no luck. When I do a tcpdump I > see that the RTP packets are sent to 0.0.0.0. How do I get the phone to > send to the * box? > > > > 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
