[Asterisk-Users] Re: PROGRESS with cause code 31 received
I have upgraded to HEAD and still, this problem persists...I can dial any other numbers through my provider with no issues, but any toll-frees do NOT work still.I have searched on Google as well, and I can't find any help..
Can anyone assist?DanaOn 11/28/05, Dana Olson <[EMAIL PROTECTED]> wrote:
I have been trying to work this problem out with my IAX provider.
I dial a toll-free number, ex: 1-888-876-6262, and I get a "due to technical difficulties" message.
I set my debug level to 9, and all I see when I dial out is this:
-- Executing Dial("SIP/27-51de", "IAX2/voctel/1766262||T") in new stack-- Called voctel/1766262-- Call accepted by
204.14.18.189
(format ulaw)-- Format for call is ulaw-- IAX2/voctel-3 is proceeding passing it to SIP/27-51de-- IAX2/voctel-3 is making progress passing it to SIP/27-51de-- Hungup 'IAX2/voctel-3' == Spawn extension (longdistance, 1766262, 1) exited non-zero on 'SIP/27-51de'
What my IAX provider sees on the other end is this:
-- Executing Dial("IAX2/tor-hub-13", "Zap/G1/1766262||g") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/1766262
-- Zap/21-1 is proceeding passing it to IAX2/tor-hub-13
-- PROGRESS with cause code 31 received
-- Zap/21-1 is making progress passing it to IAX2/tor-hub-13
-- Hungup 'Zap/21-1'
I did a search through the mailing list and in the wiki. I found that cause code "is used to report a normal event only when no other cause in thenormal class applies." and
#define AST_CAUSE_NORMAL_UNSPECIFIED 31.
I am running Asterisk 1.2.0 and I am not sure what my provider is using, some version of HEAD is all I know.
I am at a loss... I don't know the last time I tried to dial a toll-free from here, but it was working.
Can anyone help steer me in the right direction?
Thanks!
Dana
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[Asterisk-Users] PROGRESS with cause code 31 received
I have been trying to work this problem out with my IAX provider.
I dial a toll-free number, ex: 1-888-876-6262, and I get a "due to technical difficulties" message.
I set my debug level to 9, and all I see when I dial out is this:
-- Executing Dial("SIP/27-51de", "IAX2/voctel/1766262||T") in new stack-- Called voctel/1766262-- Call accepted by 204.14.18.189
(format ulaw)-- Format for call is ulaw-- IAX2/voctel-3 is proceeding passing it to SIP/27-51de-- IAX2/voctel-3 is making progress passing it to SIP/27-51de-- Hungup 'IAX2/voctel-3' == Spawn extension (longdistance, 1766262, 1) exited non-zero on 'SIP/27-51de'
What my IAX provider sees on the other end is this:
-- Executing Dial("IAX2/tor-hub-13", "Zap/G1/1766262||g") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/1766262
-- Zap/21-1 is proceeding passing it to IAX2/tor-hub-13
-- PROGRESS with cause code 31 received
-- Zap/21-1 is making progress passing it to IAX2/tor-hub-13
-- Hungup 'Zap/21-1'
I did a search through the mailing list and in the wiki. I found that cause code "is used to report a normal event only when no other cause in thenormal class applies." and
#define AST_CAUSE_NORMAL_UNSPECIFIED 31.
I am running Asterisk 1.2.0 and I am not sure what my provider is using, some version of HEAD is all I know.
I am at a loss... I don't know the last time I tried to dial a toll-free from here, but it was working.
Can anyone help steer me in the right direction?
Thanks!
Dana
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Re: [Asterisk-Users] Realtime IAX
On 9/2/05, Chris A. Icide <[EMAIL PROTECTED]> wrote: Dana Olson wrote: Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana Actually, the current CVS Head usage is rtupdate=, it was changed from rtnoupdate= not too long ago. If you are using 1.2 I'm not sure which is correct. I went through this battle of getting this to work the beginning of this week, and the four settings I listed in my last post made all the difference. -Chris Just to follow up with this thread, kpflemming provided the solution that I overlooked - the port column in the iax table was set to 0 instead of 4569. I didn't think to change it because the wiki said that the port, ipaddr, etc were all optional. For IAX peers, the port is not optional. I added a note to the wiki stating so as well. -- Dana ___ --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] Realtime IAX
On 9/2/05, Chris A. Icide <[EMAIL PROTECTED]> wrote: > I am having the exact same issues. I even tried to madk my IAX peer> account in both the database, and in the iax.conf file (with different> names, but same info) and the static one works, but not the database > one. I am using 1.2.0-beta1.>> If I specify the user:[EMAIL PROTECTED] on the dialplan, it works, but> this is bypassing the peer in the iaxpeers table in the database.>> I contacted my IAX provider, and he was not seeing the dial request > come across or anything, so where that circuit-busy is coming from, I> don't know...>> Did you ever get a resolution? Is this maybe a bug that should be> opened on the Digium tracker? >> --> DanaMake sure you have the following setting in your iax.conf file.rtcachefriends=yesrtupdate=yesrtautoclear=nortignoreexpire=yesAlso, you will still need your register => statement if you needed it before you started using realtime-Chris Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana ___ --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] Realtime IAX
On 8/5/05, Carlos Chavez <[EMAIL PROTECTED]> wrote:
I am using Asterisk CVS from last week and have been using Realtime SIPfor a couple weeks now without any problems. Yesterday I decided to turn onRealtime IAX but I am having problems dialing to my long distance providers
like Voicepulse, Sixtel or Nufone. I get the following:-- Executing Dial("SIP/2001-3761", "IAX2/[EMAIL PROTECTED]/19566680301")in new stack-- SIP Seeding peer from astdb: '2001' at
[EMAIL PROTECTED]:5060 for 3600-- Called [EMAIL PROTECTED]/19566680301Aug 5 10:25:50 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall due to slow response-- IAX2/voicepulse-11 is circuit-busy
-- Hungup 'IAX2/voicepulse-11' == Everyone is busy/congested at this time (1:0/1/0)-- Executing Dial("SIP/2001-3761", "IAX2/[EMAIL PROTECTED]/19566680301") in newstack-- Called
[EMAIL PROTECTED]/19566680301Aug 5 10:25:54 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall due to slow response-- IAX2/NuFone-2 is circuit-busy-- Hungup 'IAX2/NuFone-2' == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Dial("SIP/2001-3761", "IAX2/[EMAIL PROTECTED]/19566680301") in newstack-- Called [EMAIL PROTECTED]/19566680301-- Seeding 'pbxserver' at 66.135.38.93:4569
for 60Aug 5 10:25:58 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall due to slow response-- IAX2/sixTel-13 is circuit-busy-- Hungup 'IAX2/sixTel-13' == Everyone is busy/congested at this time (1:0/1/0)
As you can see none of them go through. I have another Asterisk serverconnected with IAX2 that does work. To that server I can dial any extensionwithout problems. I used
http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX toconfigure my * server. Any ideas? All three providers were working before Ichanged to Realtime IAX and I made sure to put all the necessary information
into the Database.--Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001
I am having the exact same issues. I even tried to madk my IAX peer
account in both the database, and in the iax.conf file (with different
names, but same info) and the static one works, but not the database
one. I am using 1.2.0-beta1.
If I specify the user:[EMAIL PROTECTED] on the dialplan, it works, but this
is bypassing the peer in the iaxpeers table in the database.
I contacted my IAX provider, and he was not seeing the dial request
come across or anything, so where that circuit-busy is coming from, I
don't know...
Did you ever get a resolution? Is this maybe a bug that should be opened on the Digium tracker?
--
Dana
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[Asterisk-Users] Re: Unresolved symbols - Zaptel 1.0.9 and Linux 2.4.31
On 7/4/05, Dana Olson <[EMAIL PROTECTED]> wrote: > I installed a vanilla 2.4.31 kernel from kernel.org and my system was > working great. > > Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols: > > > # modprobe zaptel > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol proc_mkdir_R8712438a > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol add_wait_queue_R93ee100c > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol remove_wait_queue_R8f9a6c4c > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol create_proc_entry_Re7252d3a > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol register_chrdev_Rad115a94 > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol remove_proc_entry_R9a57b3c8 > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol __wake_up_R2c77a2af > /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: > unresolved symbol __pollwait_R5ab342f6 > /lib/modules/2.4.31/misc/zaptel.o: insmod > /lib/modules/2.4.31/misc/zaptel.o failed > /lib/modules/2.4.31/misc/zaptel.o: insmod zaptel failed > > I searched the list and the previous posts didn't shed any light on my issue. > > If you need my kernel config, I can provide that, or anything else you need. > > Thanks in advance. > -- > Dana Oh FFS. I knew that as soon as I posted here I would figure it out. Apparently the -f flag for ln doesn't actually work how I thought it did, and my symlink to linux-2.4.31 didn't get updated properly. -- Dana ___ 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] Unresolved symbols - Zaptel 1.0.9 and Linux 2.4.31
I installed a vanilla 2.4.31 kernel from kernel.org and my system was working great. Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols: # modprobe zaptel /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol proc_mkdir_R8712438a /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol add_wait_queue_R93ee100c /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol remove_wait_queue_R8f9a6c4c /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol create_proc_entry_Re7252d3a /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol register_chrdev_Rad115a94 /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol remove_proc_entry_R9a57b3c8 /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol __wake_up_R2c77a2af /lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o: unresolved symbol __pollwait_R5ab342f6 /lib/modules/2.4.31/misc/zaptel.o: insmod /lib/modules/2.4.31/misc/zaptel.o failed /lib/modules/2.4.31/misc/zaptel.o: insmod zaptel failed I searched the list and the previous posts didn't shed any light on my issue. If you need my kernel config, I can provide that, or anything else you need. Thanks in advance. -- Dana ___ 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] Gizmo: Skype done right?
I think they were hoping that the client would connect to Asterisk, which makes it kinda useless, really.. But connecting Asterisk to the Gizmo network is handy. -- Dana On 7/4/05, Adrian A <[EMAIL PROTECTED]> wrote: > I have a Gizmo account working perfectly in my Xten Eyebeam, so there > should be no problem using it for Asterisk. You already have the > username (1747...etc) and your password, the proxy is > proxy01.sipphone.com (or you can sniff packets to see where SIP > messages are being sent to). ___ 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] audiocodes
On 6/29/05, Joe Murray <[EMAIL PROTECTED]> wrote: > Is anyone on this list using and audiocodes FXO gateway? I have > Asterisk(1.07 on OS X) setup and working fine, including SIP phones > and IAX2 phones - I can make outbound calls just fine and receive > inbound calls just fine. However, I can't seem to find the right > series of DTMF settings on the AudioCodes to allow DTMF tones to be > sent after an outbound call is connected(phone banking, long distance > provider etc...) while still allow the client devices(phones) to > access Asterisk voicemail. It seems I can either have the phones use > inband DTMF and work with the Audiocodes PSTN's or outband and work > with Asterisk, but not both? Any help/thoughts/experiences would be > appreciated... > > -joe I think there is the ability to set the options on the more recent firmwares (4.4 series) to allow either/or for DTMF types on the MP-108s and Mediant 2000 devices. I don't know exactly what or where the settings are, but be sure you've got the most recent firmware you can. They are generally better. I am not exacly using the AudioCodes devices so much with Asterisk as with SER, so perhaps the settings I saw wouldn't help you out at all. ___ 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] Pass variable to Authenticate?
I'm trying to figure out a way to make my own agent login, because I don't like how the default works. I have the login and logout working fine using the dynamic add and remove commands, but I need to be able to create a list of users and passwords. I thought of a way to do it using a list of passwords, but the agent would only ever be prompted for their password. I won't want that. Basically, what I want to do is this: Read in the username, ex. 4567 Read in the password, ex. 1234 Pass 12344567 to the Authenticate command, which checks in the file for that password. I can't figure out a way to do it, and it may not even be possible. Is there another way to do it? Can I somehow utilize the DBget and DBput commands? I couldn't find much info on it in the wiki, and my impression is that I have to use only the blacklist or cidname databases. Any ideas would be helpful. -- Dana ___ 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] Digium MOH
On 5/3/05, Matt Riddell <[EMAIL PROTECTED]> wrote: > Chris Mason wrote: > > Why not? > > Because you have not licensed the file for broadcasting across your > telephone network. Don't jump to conclusions. ;) > How many other people are there here that write music? Would there be > any interest in creating a pool of music for Asterisk? I've thought about it... Perhaps I would be interested in contributing a song or two. > Would there be any chance of creating a GPL exception for them if we > donated them? > > I have rather a few songs, mostly in the trance/psytrance genre but also > dub and DnB. I write metal and industrial and all inbetween, which isn't that much. But I could come up with something that isn't so harsh, specifically for Asterisk. I'm a lame coder, so maybe my music skills would be a better way of contributing... > Ideas? Well, we could put up a website or a project on SourceForge or something perhaps and put any donated music there? I don't remember what SF's rules are, but there has to be something we could do, assuming that the music wasn't simply accepted into Asterisk. It could be in an add-on archive, kinda like the additional sound files pack. I dunno, but I'd contribute a song if something formed.. Let me know. -- Dana ___ 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] Recording in a call center
Wouldn't introducing Samba into the mix be even worse?
I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.
If it's such a critical system, it should have the power to do that,
but that's not the point... If I had such a critical system, I'm not
so sure that I would be saving files in real-time over the network via
Samba.
My question is, what's the difference between writing to the local
disk and over the network? What will happen if the network link goes
down? I've had bad experiences with Samba and NFS both, as far as
connectivity issue handling is concerned.
--
Dana
On 4/29/05, sjaak imap <[EMAIL PROTECTED]> wrote:
>
>
> You need something like this ??
>
> exten => _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
> exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
> exten => _0.,3,Dial,SIP/[EMAIL PROTECTED]
>
> and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
>
> That would be the job.
>
>
> Sjaak
>
>
> > I would like to record two months of calls. The call center does not
> > have a huge volume, probably like 60 calls a day and average about 15
> > min a call. I am using a quad port e1 card from digium. i would like
> > to record the calls on a seperate server than the one running asterisk
> > to avoid any problems.
> >
> > any ideas?
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Re: [Asterisk-Users] Recording in a call center
60 calls a day is nothing. I'm sure your Asterisk box can handle it with the standard Monitor command. I've recorded many calls, 8+ hours straight and I'm on a crap old Pentium 3 633MHz system. What exactly do you fear will happen if you record on the Asterisk box? -- Dana On 4/29/05, Steve Totaro <[EMAIL PROTECTED]> wrote: > I would like to record two months of calls. The call center does not have a > huge volume, probably like 60 calls a day and average about 15 min a call. > I am using a quad port e1 card from digium. i would like to record the > calls on a seperate server than the one running asterisk to avoid any > problems. > > 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 > > ___ 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] Queue Monitor Filename Problem
On 4/29/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi ! > I am using queues with MOnitor Application but the thing is that Iwant to > save the files starting with the Answering agent name. I have tried a lot > of things but nothing seems to work. If i put Monitor application on top > of dialing the agent then as soon as agent picks up the recording hangs up > without recording anyhting. And if I put the Monitor application on top of > Queue command then I have to specify the saving filename before I know > that to which agent the call is going. ANy comments , suggestions > appreciated. > Thanks, > Usman. You could start the recording with a manager command remotely via telnet... I've been working on this, but my problem is that I don't know socks and PHP too well. ___ 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] Confused on G723 and G729
You would need a transcoding license between the Asterisk PBX and the G711 phone... On 4/28/05, Matt <[EMAIL PROTECTED]> wrote: > So > > > [g729 provider] -(SIP or IAX)--- [g729 asterisk server] > > This is how I'd be setup.. actually more like this: > > [g729 provider] --(sip) [g729 asterisk > server](sip)---[g711 sip phone client]. > > So... if I understand this correctly.. I *would not* for *any* reason > need a license going from the g711 client since to voicemail/etc > is fine.. and going out to the provider is not in my system? > > However, if someone calls IN from the g729 provider and wants to check > voicemail, then I'd need a license? > ___ > 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] RJ45 to RJ11?
On 4/27/05, Paul Shiflet <[EMAIL PROTECTED]> wrote: > I just received my TDM400 card from digium with 2 fxo and 2 fxs > interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS > phones. How do i interface my POTS phones with this; can i just crimp an > RJ45 connection on the end of the phone cord? > > Paul Plug the RJ11 connectors into the RJ45 jacks... ;) ___ 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 a peer over the asterisk manager with a php script
On 4/27/05, Guy Boehm <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I want to call a peer over the Asterisk Manager with this php-script:
>
>
>
>
>
> $socket = fsockopen("192.168.204.44","5038", $errno, $errstr,
> $timeout);
> fputs($socket, "Action: Login\r\n");
> fputs($socket, "UserName: test\r\n");
> fputs($socket, "Secret: test\r\n\r\n");
> //fputs($socket, "Action: ListCommands\r\n\r\n");
>
> fputs($socket, "Action: Originate\r\n");
> fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");
> fputs($socket, "Exten: 1009\r\n\r\n");
> fputs($socket, "Context: test\r\n\r\n");
> fputs($socket, "Priority: 1\r\n\r\n");
>
>
> fputs($socket, "Action: Logoff\r\n\r\n");
> while (!feof($socket)) {
> $wrets .= fread($socket, 8192);
> }
> fclose($socket);
> echo << ASTERISK MANAGER OUTPUT:
> $wrets
> ASTERISKMANAGEREND;
> ?>
>
>
>
>
> I got this resulat:
>
> ASTERISK MANAGER OUTPUT:
Asterisk Call Manager/1.0
Response:
> Success
Message: Authentication accepted
Response: Error
Message: Invalid
> channel
Response: Error
Message: Missing action in request
Response:
> Error
Message: Missing action in request
Response: Error
Message: Missing
> action in request
Response: Goodbye
Message: Thanks for all the fish.
I
> tried many diffrent SIP/Channels but nothing works
>
> THX
On top of what Richard mentioned, you have too many \r\n in the
Originate action parameters. Only the final one should have two sets,
the others hould have only one.
--
Dana
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Re: [Asterisk-Users] Phone Recommendation.
On 4/27/05, Sean A. Newton <[EMAIL PROTECTED]> wrote: > On Tue, 26 Apr 2005, Dana Olson wrote: > > > You mean like the problem I described earlier on this list? > > > > http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html > > > > I am not sure why I didn't think of disabling call waiting, but that > > seemed to work with a Grandstream BudgeTone phone... I'm doing more > > testing now. > > Sounds exactly like the same problem. Of course, the $65 grandstreams > allow you to disable call waiting.. The stupid $130+ Polycom's don't. :( Heya. I did a bit of testing, and the $175 snom 190 does work properly once I disabled CW too. Woo! That's great news. Sucks about Polycom though. I was advised to stay away from them, as they apparently have a bad rep... But I don't know anything about 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] VOIP Gateways & Asterisk
Another option is the AudioCodes Mediant 2000 devices. I have one with 8 T1/E1 ports, one with 4 ports, and one with 1 port. You can also get up to 16 ports, and there is a 2-port model as well. I am using SIP with them, but I think you can use H.323 with a different firmware. I am using them in a different application than you, but they are standard SIP, and they seem to work great. Just another choice. -- Dana ___ 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] Phone Recommendation.
On 4/26/05, Sean A. Newton <[EMAIL PROTECTED]> wrote: > On Mon, 25 Apr 2005, Wiley Siler wrote: > > > Call waiting can be disabled in Asterisk via *71 regardless of the phone > > used. > > > > Cheers, > > Wiley > > Well, this is part of a larger problem I'm having. > > I can't get CheckGroup/SetGroup to work as I think it should for my > dynamically added ACD agents. > > The management here is frustrated, and they just want to buy a few phones > that simply can have call waiting disabled. > > --Sean You mean like the problem I described earlier on this list? http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html I am not sure why I didn't think of disabling call waiting, but that seemed to work with a Grandstream BudgeTone phone... I'm doing more testing now. ___ 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] Grandstream ATA 286 problems
I've had an issue with my 286 ever since I got it. Basically, the web interface doesn't load, and I can't make any calls - although I get dialtone. Also, I can call it and it will ring. But I get no audio. The main issue is that I can't get into the web interface anymore... I did once, but not anymore. I contacted the vendor I bought it from, and they said to contact Grandstream. I contacted Grandstream, and they told me to hit refresh in my browser After sending them the Ethereal trace, I haven't heard back from them yet. I think it's the worst purchase I've ever made. On 4/25/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Anobody had any problem with GS ata 286? The past few days Ive been having > some problem with it, while making a call or during a call, I suddely hear a > low noise like a car engine starting and then the ata dies, as if it got > stuck or frozen. > > Anybody had these problems? ___ 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] can't make my PRI dial out
Did you try dialing out over ZAP/g1?
On 4/22/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
> I have a full PRI installed on my * machine. I can get inbound calls
> just fine but can't make outbound ones.
>
> Zaptel.conf says;
>
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
>
> zapata.conf says
>
> language=en
> context=default
> switchtype=4ess
> pridialplan=unknown
> signalling=pri_cpe
> channel=>1-23
> echocancel=yes
> group=1
>
> dial string in extensions.conf says
>
> ; calls to the outside world via the PSTN
> exten => _81NXXNXX,1,Dial(ZAP/1/${EXTEN:1})
>
> When I try to dial a number I get
>
> - Executing Dial("SIP/3710-23ea", "ZAP/17327356701") in new stack
> Apr 22 10:19:17 NOTICE[28197]: app_dial.c:803 dial_exec: Unable to
> create channel of type 'ZAP' (cause 0)
> == Everyone is busy/congested at this time
>
> pri show span 1 says
>
> 120b-pbx*CLI> pri show span 1
> Primary D-channel: 24
> Status: Provisioned, Up, Active
> Switchtype: AT&T 4ESS
> Type: CPE
> Window Length: 0/7
> Sentrej: 0
> SolicitFbit: 0
> Retrans: 0
> Busy: 0
> Overlap Dial: 0
> T200 Timer: 1000
> T203 Timer: 1
> T305 Timer: 3
> T308 Timer: 4000
> T313 Timer: 4000
> N200 Counter: 3
>
> Paid support from Digium sucks royally. They don't even know what their
> own error codes mean!!
>
> Any ideas?
>
> Thanks
>
> Mark
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[Asterisk-Users] Dynamic queue member behaviour
I found that if I dynamically add, for example SIP/8000, to a queue, then calls in the queue will sorta pile up on the 9 extensions on that phone - not what we want to happen. If I log in to the queue using AgentLogin, then the behaviour is as expected - one call at a time. Is there a way around this, or am I adding dynamically to the queue incorrectly? Thanks in advance for any assistance. ___ 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] leastrecent queue option
For agent queues, I checked this option out: ; leastrecent - ring interface which was least recently called by this queue When I use this, if I have 3 agents logged into the queue, and I pump in 3 calls to the queue, I would expect that one call would go to each agent, but instead, it doesn't work this way. All calls ring on 3 extensions of one phone only. If I take the call on the phone, then hang it up, any further calls will go to the next phone. The problem here, to me, is that when the agent's phone rings, the pointer to which agent was least recently called doesn't get reset to the next agent, as one might expect it to... At the very least, I would think that it would reset at the pickup of the call, if not at the ring. Am I wrong, or does anyone know if this is fixed in CVS? Even if it is, I'm not planning on using a development version of Asterisk in production, so CVS is not an option. I will continue testing various queue strategies until I find one that is useable, but any insight into this particular behaviour is appreciated. -- Dana ___ 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] Recommended Linux Dist. for Asterisk
I'm using Debian Stable both at home and at work, and Asterisk runs fine for me. Use what you know. -- Dana ___ 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] Monitor via Manager question
Well, I guess that I'm not as good as I once was... If anyone would care to assist me in this, I would appreciate it. If you want to contact me even on IM, IRC or off-list, anything's cool with me... I would really appreciate the help. -- Dana ___ 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] Monitor via Manager question
Alright, thanks you guys. I was hoping to not have to do that, but I guess it's time to get my PHP on. I find myself re-learning it every time I start a new project. I love the language, I'm just forgetful. :-) ___ 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] Monitor via Manager question
Hello. I checked in the wiki and read a bunch of old threads from this mailing list but haven't found what I'm looking for. I'm using a simple PHP script, and here is the relevant portion: fputs($socket, "Action: Monitor\r\n"); fputs($socket, "Channel: Zap/1-1\r\n\r\n"); That works fine. As does this: fputs($socket, "Action: Monitor\r\n"); fputs($socket, "Channel: SIP/8000-h4d8\r\n\r\n"); But what I need to be able to do is this: fputs($socket, "Action: Monitor\r\n"); fputs($socket, "Channel: SIP/8000\r\n\r\n"); And have it record either the first call that is up on SIP/8000, or the last, or whatever (doesn't matter, only one call at a time will be up on this line). However, if I try this, it always comes back to me with: Response: Error Message: No such channel Because I am not specifying the actual call that is up. Is there any way to do this? Or can I somehow easily look up what Zap channel is used by SIP/8000 and pass that? The other twist is that SIP/8000 will be specified by a variable passed through a form. Basically, I want a web form with two buttons and a text box: Start Rec., Stop Rec., and User Ext.. I didn't start out that complex though, just right now it's a simple PHP script, and it was taken from the Wiki. I need to get the core functionality working properly before I add the buttons and whatever. Thanks in advance for any advice. -- Dana ___ 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: Starting with Asterisk-SIP
On 4/19/05, ruben cuevas rumin <[EMAIL PROTECTED]> wrote: > I would like to know if for this simple test (communication using IP > address directly) , need I a dialplan or no??? And if I need a > dialplan, where I could obtain any example of a extension.conf file > for this simple test. (because I only find examples for other more > difficult implementations). If you're dialing direct, you don't need Asterisk at all. You can just dial peer-to-peer SIP... ___ 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] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?
On Apr 12, 2005 9:38 AM, Chuck Bunn <[EMAIL PROTECTED]> wrote: > Hi, > > Actually I guess what I am looking for is semi-sealed box that I can add > 1 or 2 PCI cards too. A regular PC work work in most cases since I do > not want a keyboard or mouse attached to it. I do not want users > screwing with the system. If it is sealed with no monitor/keyboard/mouse > then they can't screw it up very easily. I guess I am looking for > something that is somewhere in between a PC and Linksys router box. One > possibility might be a thin client box, but I haven't found any sources > for an OEM box. I looked at the HP > (http://h18004.www1.hp.com/products/thinclients/index_t5000.html) thin > clients but I can get a Dell Box for the same price that does more. > > Thanks Have you checked eBay? http://lists.digium.com/pipermail/asterisk-users/2005-April/100861.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
Re: [Asterisk-Users] Dialing Out
On Apr 11, 2005 8:11 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > nat=no > disallow=all > allow=g729 > allow=g726 > auth=plain > context=default > canreinvite=yes > username=USERNAME > secret=PASSWORD > dtmfmode=info > fromdomain=REALM > fromuser=USERNAME > qualify=1000 > insecure=very > > I am using Asterisk 1.0.7 compiled into RPMs from the tarball running on > CentOS Enterprise 4... > > Can someone point me in the right direction... > > Doug You haven't stated what your problem is. ___ 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] Sangoma VS. Digium
On Apr 7, 2005 6:20 AM, Matteo Brancaleoni <[EMAIL PROTECTED]> wrote: > Sangoma doesn't do that. they don't sell directly, thus allowing > resellers to have a money gain and pay the time to support the end > user. Actually, they do sell directly. I emailed them a short time ago, and they gave me their price. They also invited me to go visit their offices. I made it clear that I was an end user, and not a reseller. -- Dana ___ 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] Sangoma VS. Digium
On Mar 31, 2005 1:44 PM, Dana Olson <[EMAIL PROTECTED]> wrote: > On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > > > My understanding is that to an extent when we buy Sangoma > > > > > we're putting the dagger to Digium. > > > > > > > > If anything "puts the dagger" to Digium it'll be their own inability to > > > > engineer reliable hardware. > > > > > > > > I appreciate what Digium has done for Asterisk, but reliability > > > > expectations > > > > for phone equipment are extremely high. I sympathize with people who > > > > need > > > > hardware that doesn't need to be restarted once a week just to do its > > > > job > > > > properly. If Digium can't deliver on those reliability expectations, > > > > and do > > > > it soon, people are going to switch to companies that can. And you know > > > > what? I don't blame them. > > > > > > > > > The Digium boards need to be restarted once a week? > > > > > > Please clarify this. I was dead set on getting in a Sangoma A104 for a > > > production Asterisk box, but then I read this thread and felt that it > > > didn't matter so much what I would order... And so I was deciding to > > > stick with Digium. And then I read your scary comment. > > > > > > I've currently got a Digium board filled with 3 T1s, but it hasn't > > > been under heavy use right yet, due to my attention being pulled from > > > * and put onto SER+AudioCodes devices for other applications, and I > > > haven't had to restart yet. Is this going to change? What's the deal? > > > > > > Please clarify your statement for me, as I need reliability as well. > > > > I'll jump in here (but I'm not the original poster). The "once a week" > > thing relates to the digium TDM card (fxo and/or fxs modules). I don't > > believe the T1 cards are an issue that requires driver reloads. > > Alright, that helps clarify it a bit, but then again, I have been > running Asterisk at home with a TDM card for a couple months and > haven't had to restart it for a long time. Is it a requirement or just > simply a recomendation? I shouldn't have said anything. My incoming pots line stopped responding this weekend. I found out when I got an email from someone telling me that it just keeps ringing and ringing. This never happened before... Strange thing is though, today, my IAX number wasn't responding. It's like I'm silently losing my registration or something. Totally unrelated, I know. I'm gonna go back to 1.0.6 because I ran it for a good while with no problems. I hope this solves it. It fixed the issues I had with 1.0.7 at work. -- Dana ___ 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-1.0.7 Build - Serious issues
On Mar 31, 2005 1:24 PM, Dana Olson <[EMAIL PROTECTED]> wrote: > On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT) > <[EMAIL PROTECTED]> wrote: > > Folks! > > > > I want to let everyone know that I have been trying to migrate from > > 1.0.6 to 1.0.7 last few days and I have come across serious issues in > > the build 1.0.7. What I found are listed below. I would recommend > > everyone to hold off any upgrade till the next build. > > > > 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the > > Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say > > extension numbers for the called user. My guess is all these .gsm files > > are corrupt and hence you don't hear anything. > > > > 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are > > corrupt. When we tried to play these files using a media player, all we > > hear is gibberish. > > > > 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we > > configure this for RFC2833. > > > > Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been > > able to find a fix? > > > > Seshu > > Just to add to this, I've been having some issues with audio with > 1.0.7 as well. I haven't yet downgraded back to 1.0.6 to see if it > solves it, but basically, I'm hearing some artifacts, and these didn't > occur in the last 3 or 4 builds. I'm using the same phone and same > config as I had been on 1.0.6. Other people that I call say that my > phone sounds like crap now too. Also, when dialing over a Zap channel, > the audio seems to sorta stutter now, at the very first second or two > of a call. This didn't happen prior to 1.0.7. > > When I get some time that the server is not in use, I will downgrade > and try to confirm that it is definitely an issue with the new build, > but so far it seems that way. > -- > Dana I rolled back to Asterisk 1.0.6 this morning and things seem back to normal as far as voice quality goes. -- Dana ___ 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] Xten-lite for linux
On Mar 31, 2005 8:55 PM, Bruno Hertz <[EMAIL PROTECTED]> wrote: > Brian Capouch <[EMAIL PROTECTED]> writes: > > > Hmmm. I just got the latest beta build, which identifies itself as 1105d. > > > > The keypad functionality is perfect. > > Hmmm. Good for you. We were talking about sjphone, though :) > > Regards, Bruno. I'm pretty sure that I used SJphone to check my VM. I'll test again. But there is a new beta out on their site (and it's newer than the Windows build). Maybe they added a dialpad? -- Dana ___ 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] Xten-lite for linux
On Thu, 31 Mar 2005 14:49:11 -0300, Carlos Gabriel Drach <[EMAIL PROTECTED]> wrote: > Kris Edwards wrote: > > >This is the best linux sip phone I've used so far. Audio quality has > >been perfect and it seems really stable, so hopefully it will be out of > >beta soon. > > > >I might actually pay for the full version! (not counting console games, > >that would be the second piece of software I've purchaced since 1987). > >___ > >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 > > > > > > > Where can i get that version? > > Not found any link on xten site... > > Thanks Sign up for their forums and then email them ([EMAIL PROTECTED] I think) with a request to join the Linux beta. ___ 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-1.0.7 Build - Serious issues
On Thu, 31 Mar 2005 10:01:29 -0600, Henry Devito <[EMAIL PROTECTED]> wrote: > An additional fault in 1.0.7 When you log into voicemail and select advanced > options there are none. On previous versions it would ask if you would like > to send a message, etc. That's strange man. My VM works fine on 1.0.7. -- Dana ___ 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] Xten-lite for linux
> > Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends > > from my disgust with everything else. In particular, kphone, and > > sjphone. I have noticed latency with xten in meetme, but if I just dial > > somebody it works better than anything I've tried (so far.. I've only > > spend about 1 hour talktime). Anyway, I'm certainly more hip on open > > source, and can't wait to try gnomemeetings sip once I can actually get > > it to compile :/ > > > > I have not tried lipz4 yet either (not sure if it will work w/ gentoo, > > but I might give it a try if I can find any rave reviews) > > Funny how experiences vary. E.g. I thought sjphone for linux wasn't too > bad, if it only had a dial pad. > > Anyways, I'm trying the latest xlite beta right now, and I must say it really > has improved. I've been sticking to gnomemeeting yet, but here seems to > be a candidate to be taken into serious consideration for everyday use. > Especially since the GM/SIP support apparently takes it's time. > > Regarding my previous statements, echo tests with a local and a public * > server are pretty fine now, and audio/latency is way better compared to my > last tests maybe two months ago. In this respect, the current beta actually > does equally well as GM and could be considered fit for production. > > What I personally don't like though is that funky interface where one even > can't always be sure where the mouseclick 'hot spots' are. For an mp3 player > this might be OK, but this being a tool one is supposed to really work with > I'm not sure what's going through the mind of those people. Hopefully, a > reasonable skin comes up some time in the near future ... > > Still, thanks (finally) for reminding me of a phone that I've put aside maybe > a little too early :) > > Regards, Bruno. What's wrong with using your keyboard's Num pad? ___ 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] Sangoma VS. Digium
On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > > My understanding is that to an extent when we buy Sangoma > > > > we're putting the dagger to Digium. > > > > > > If anything "puts the dagger" to Digium it'll be their own inability to > > > engineer reliable hardware. > > > > > > I appreciate what Digium has done for Asterisk, but reliability > > > expectations > > > for phone equipment are extremely high. I sympathize with people who need > > > hardware that doesn't need to be restarted once a week just to do its job > > > properly. If Digium can't deliver on those reliability expectations, and > > > do > > > it soon, people are going to switch to companies that can. And you know > > > what? I don't blame them. > > > > > > The Digium boards need to be restarted once a week? > > > > Please clarify this. I was dead set on getting in a Sangoma A104 for a > > production Asterisk box, but then I read this thread and felt that it > > didn't matter so much what I would order... And so I was deciding to > > stick with Digium. And then I read your scary comment. > > > > I've currently got a Digium board filled with 3 T1s, but it hasn't > > been under heavy use right yet, due to my attention being pulled from > > * and put onto SER+AudioCodes devices for other applications, and I > > haven't had to restart yet. Is this going to change? What's the deal? > > > > Please clarify your statement for me, as I need reliability as well. > > I'll jump in here (but I'm not the original poster). The "once a week" > thing relates to the digium TDM card (fxo and/or fxs modules). I don't > believe the T1 cards are an issue that requires driver reloads. Alright, that helps clarify it a bit, but then again, I have been running Asterisk at home with a TDM card for a couple months and haven't had to restart it for a long time. Is it a requirement or just simply a recomendation? ___ 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] Sangoma VS. Digium
Do you think Digium has enough money for that? I don't know how large Sangoma is, but they've been around almost as long as I have... I think it'd be a great idea though, if they have the cash for it.. I bet it'd pay off too. On Thu, 31 Mar 2005 11:00:48 -0500, mattf <[EMAIL PROTECTED]> wrote: > Here's an idea, Digium buys Sangoma with the massive amounts of cash they > are getting from venture capitalists and just integrate Sangoma designs into > their boards. Not sure how Sangoma would feel about this idea though. > > MATT--- > > > -Original Message- > From: Matthew Boehm [mailto:[EMAIL PROTECTED] > Sent: Thursday, March 31, 2005 10:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Sangoma VS. Digium > > Brian Capouch wrote: > > > I'll be glad to stand corrected, but if that assertion is in fact > > true, we should be careful to do things that actually damage Digium's > > ability to leverage their development of Asterisk with their hardware > > sales. > > It sucks that its such a fine line. On the one had, it is good to have > competition. Keeps prices in check, and gets new features out faster. > > But on the other hand, yes, buying from someone else may say to Digium > "well, I guess we can stop now that they are buying someone elses cards." > > -Matthew > > ___ > 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] Xten-lite for linux
I've been meaning to try it again. A large number of builds have been sent since I last tried. And boy, it was sooo slow and more resource-intensive than its Windows counterpart. I haven't been using a softphone at home because I'm waiting for GnomeMeeting w/SIP to get into Ubuntu or Debian. Instead, I just use a cordless phone plugged into my TDM card. -- Dana On Thu, 31 Mar 2005 16:57:22 +0200, Bruno Hertz <[EMAIL PROTECTED]> wrote: > "hank smith" <[EMAIL PROTECTED]> writes: > > > do you know if it is gtk2? > > It appears to be: > > $ ldd xlite-linux-22 > ... blah ... > libgtk-x11-2.0.so.0 => /usr/lib/libgtk-x11-2.0.so.0 > ... blah ... > > 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 > ___ 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-1.0.7 Build - Serious issues
On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote: > Folks! > > I want to let everyone know that I have been trying to migrate from > 1.0.6 to 1.0.7 last few days and I have come across serious issues in > the build 1.0.7. What I found are listed below. I would recommend > everyone to hold off any upgrade till the next build. > > 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the > Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say > extension numbers for the called user. My guess is all these .gsm files > are corrupt and hence you don't hear anything. > > 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are > corrupt. When we tried to play these files using a media player, all we > hear is gibberish. > > 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we > configure this for RFC2833. > > Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been > able to find a fix? > > Seshu Just to add to this, I've been having some issues with audio with 1.0.7 as well. I haven't yet downgraded back to 1.0.6 to see if it solves it, but basically, I'm hearing some artifacts, and these didn't occur in the last 3 or 4 builds. I'm using the same phone and same config as I had been on 1.0.6. Other people that I call say that my phone sounds like crap now too. Also, when dialing over a Zap channel, the audio seems to sorta stutter now, at the very first second or two of a call. This didn't happen prior to 1.0.7. When I get some time that the server is not in use, I will downgrade and try to confirm that it is definitely an issue with the new build, but so far it seems that way. -- Dana ___ 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] Sangoma VS. Digium
On Thu, 31 Mar 2005 10:00:12 -0500, David Brodbeck <[EMAIL PROTECTED]> wrote: > > -Original Message- > > From: Brian Capouch [mailto:[EMAIL PROTECTED] > > > My understanding is that to an extent when we buy Sangoma > > we're putting the dagger to Digium. > > If anything "puts the dagger" to Digium it'll be their own inability to > engineer reliable hardware. > > I appreciate what Digium has done for Asterisk, but reliability expectations > for phone equipment are extremely high. I sympathize with people who need > hardware that doesn't need to be restarted once a week just to do its job > properly. If Digium can't deliver on those reliability expectations, and do > it soon, people are going to switch to companies that can. And you know > what? I don't blame them. The Digium boards need to be restarted once a week? Please clarify this. I was dead set on getting in a Sangoma A104 for a production Asterisk box, but then I read this thread and felt that it didn't matter so much what I would order... And so I was deciding to stick with Digium. And then I read your scary comment. I've currently got a Digium board filled with 3 T1s, but it hasn't been under heavy use right yet, due to my attention being pulled from * and put onto SER+AudioCodes devices for other applications, and I haven't had to restart yet. Is this going to change? What's the deal? Please clarify your statement for me, as I need reliability as well. 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] why even use SIP
On Sat, 26 Mar 2005 04:14:54 +0200, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Wed, Mar 23, 2005 at 04:37:02PM -0500, Dana Olson wrote: > > > My company has thousands of entries in the DHCP server, and it would > > take forever to go through each and every one of them. Not to mention > > that I, being in the telecom division, do not have access to the DHCP > > servers. > > scan for a MAC address? > > ping all the addresses in the range and then > > /usr/sbin/arp -n |grep -i that_mac_addr > > The scanning part could be done using something like: > > nmap -sP 192.168.1-5.* > > Another simple trick (assuming a mostly windows network) is to simply > ping to the broadcast address. Linux-es and macs tend to respond to > those pings and so are most devices. Windows tend to ignore those pings. > > -- > Tzafrir Cohen | New signature for new address and | VIM is > http://tzafrir.org.il | new homepage | a Mutt's > [EMAIL PROTECTED] || best > ICQ# 16849755 | Space reserved for other protocols | friend The MAC addresses are not labeled on the units. I swear I said that already. ___ 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] why even use SIP
On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin <[EMAIL PROTECTED]> wrote: > xlite doesn't seem to have this problem. X-Lite doesn't support IAX. ___ 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] Newbie pointers
On Thu, 24 Mar 2005 12:45:58 +0100, Fred Blaise <[EMAIL PROTECTED]> wrote: > Hello all > > I have come to Asterisk with no previous telco experience. > As I will be playing with Asterisk really soon, I would like to have > some pointers as to some tutorials in telco that could help me get into > all this. I am quite a beginner, don't forget :) > > Thanks a lot! > > Best, > fred Everything that everyone already said is good advice. If you a PDF document on installing Debian Stable, I would be happy to provide one for you. Just email me off-list. What I did was just installed Asterisk Stable, install the sample config files, and then started Asterisk. I just looked at the example configs and modified them. They are a great source of knowledge as well, in the commented sections. Also, just to echo what has already been said, the wiki is a great reference. Learn how to search through it and you'll be all set. I too had very limited telecom experience prior to checking out Asterisk, but it has got my interest for sure, and I've learned quite a lot. -- Dana ___ 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] Newbie Voicemail Question
On Thu, 24 Mar 2005 09:34:48 -0600, Art Zemon <[EMAIL PROTECTED]> wrote: > Folks, > > Please forgive my ignorance. I think that what I am asking must be so > obvious that no one bothers to write it down. But I don't know the > answer so... > > I want to set up * with one incoming VOIP phone number. If someone calls > me and is talking to me on that phone number, how does a second caller > get to * voicemail instead of a busy signal? > > Thanks, >-- Art Z. This is how it works on my home Asterisk setup. If you are not experiencing this, can you see if that second call is even hitting your server? I have a feeling that your VoIP provider is limiting you to a single call. Either that, or you don't have your dialplan set up properly. Might wanna paste the relevant portion here as well. -- Dana ___ 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] why even use SIP
On Thu, 24 Mar 2005 15:34:26 -, Giles Coochey <[EMAIL PROTECTED]> wrote: > > > > > > > > > > > How about scanning for it's mac address? > > http://ipscan.sf.net/ipscan.exe > > > > > > -- > > > http://www.umich2.com > > > > > > Digium doesn't label the MAC address on the device, unless it's such a > > fine print that no one can read it. I believe this has been said a few > > times in the conversation. > > Connect it with a cross-over ethernet cable to a Linux box and run > tcpdump on the Linux box, before long you'll see the IP address come up > on the tcpdump logs. Don't power it off, you want it to have an existing > DHCP lease. > > If you don't see any traffic, try making a call. Once you have the IP > you can put it back on the normal network and configure it. I know how to work around these limitations already. My point is that this is not an enterprise-ready solution. If I order 1000 of these for our IT staff, I have to go through each and every one with a crossover cable just to find the IP? Why would we bother when there so many other devices that don't have any of the flaws of the IAXy? Of course they are SIP-only, so that's the answer to the question of "why use SIP at all." Because there is no good solution for IAX yet. With a little work, the IAXy can become a product not only for hobbyists but for the corporate world as well. Until then, we will need to rely on Sipura, Grandstream, and the like for devices that can be much easier provisioned, either by keypad entry on the device itself, TFTP config files, or an HTTP interface, that support DNS name resolution, G729/iLBC/GSM codecs, have their MAC addresses labeled on them, etc. This is for my company only. Perhaps yours isn't so large and you have the time and desire to go through this process for every device in your organization, but we don't. Yes, for home users who run Asterisk, it's fine, except if they want to take the IAXy on the road with them and they don't have a static IP address. For internal use in a small company, yeah, the IAXy may be a fine solution. But when you're looking at purchasing hundreds of devices at a time, I don't think this is a good product at this time. All of that said, I like the IAXy, and I will gladly recommend buying it if you're not in my position, or if Digium develops it further to address these issues. ___ 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] why even use SIP
On Thu, 24 Mar 2005 08:09:19 -0700, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Dana Olson wrote: > > >On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit <[EMAIL PROTECTED]> wrote: > > > > > >>>6) Configuration requires Linux, as opposed to a web browser or > >>>something more standard. > >>> > >>> > >>I compiled iaxprov on Cygwin, works nicely. There's somebody on this > >>list that made a Windows version to provision it. Works nicely, GUI > >>interface, can even scan the LAN to find IAXy. Here is the link to it > >>: http://dacosta.dynip.com/asterisk > >> > >> > > > >Thanks for that link, I'm gonna try it. The main issue here is that > >this is a large company and I don't have access to the DHCP servers, > >and therefore can't just find out the IP address of this thing. > >There's another feature request. Let me dial ### or something to find > >my IP... > > > > > How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe > > -- > http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. ___ 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] why even use SIP
On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit <[EMAIL PROTECTED]> wrote: > > So how am I going to provision the device in the first place, to be > > able to dial this extension, if I don't even know the IP? > Oups, sorry, didn't think about this one. > > Check winiaxyprov, the version 1.01 can scan your network to find > IAXy. Now the only thing we need is for Digium to write the MAC > address on the device before sending it in the open world. Because if > you have more than one on your network, you can't really know which > one you need to provision. > > hth Yeah, I found that app earlier in the thread and thanked whoever it was (maybe you, can't remember) for linking to it. It's handy, as I had no way to determine the MAC or IP address prior to this, my IAXys kinda sat on the shelf collecting dust. (I did bring one home and plugged it into my Linksys router, but that's hardly an option in a large IT organization with many IAXys.) My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. Luckily I actually have a Windows desktop here at work. I'd like a scanner like that for Linux though. Maybe it's possible with some other kind of application? Anyhow, I still think it wouldn't kill them to add an IP address feature or something (an alternative would be to allow the iaxyprov tool to provision by MAC or IP, and yes, start labeling the devices with their MACs). To me, it just doesn't seem like a product that was really ready for release yet. I think it could be really great after a bit of development though, and wouldn't discourage Digium from doing so, but for now, our company can't really use these for many applications. -- Dana ___ 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] why even use SIP
On Wed, 23 Mar 2005 12:55:45 -0800, Sys Admin <[EMAIL PROTECTED]> wrote: > so seems like the verdict is go IAXy with a IAX only network ? Most of > the problems of the IAXy device seems like will be fixed with firmware > updates and wont require a hardware update.. > > this way we get the advantage of a Hardphone (human factor, just feel > good to talk on a real phone) with all the goodies of the IAX > protocol. > > t Has Digium said that they will fix the issues most of us have with the IAXy? I haven't seen it, but maybe I missed the message? If I were you, I'd get one IAXy device in and test it first, see if it is what you want... And for comparison, grab a low-end Sipura ATA in as well, see what you decide. ___ 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] why even use SIP
On Wed, 23 Mar 2005 13:59:07 -0500, Time Bandit <[EMAIL PROTECTED]> wrote: > > There's another feature request. Let me dial ### or something to find > > my IP... > That's not something to do with the IAXy, you can make an AGI script > that will tell you your IP. I had this script somewhere but I can't > find it at the moment. This would not only be valid for the IAXy, but > for any phone connected to asterisk (well, except analog phones) > > If I find it, I'll let you know. But I'm confident that somebody on > this list has something like this. > You dial some extension that call this script and it tells you your IP > using SayDigits. > > hth > So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? ./iaxyprov Usage: provision ___ 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] why even use SIP
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit <[EMAIL PROTECTED]> wrote: > > 6) Configuration requires Linux, as opposed to a web browser or > > something more standard. > I compiled iaxprov on Cygwin, works nicely. There's somebody on this > list that made a Windows version to provision it. Works nicely, GUI > interface, can even scan the LAN to find IAXy. Here is the link to it > : http://dacosta.dynip.com/asterisk Thanks for that link, I'm gonna try it. The main issue here is that this is a large company and I don't have access to the DHCP servers, and therefore can't just find out the IP address of this thing. There's another feature request. Let me dial ### or something to find my IP... > > Let's face facts there, the IAXy sucks by any definition. > No it doesn't. Granted it has a couple shortcomings, but nothing that > bad. If Digium can fix the most important ones and find a way to drop > the price a bit, this would be a great little device. You just admitted that without the features, it's not great... Seems like we're all on the same page here. No sense arguing about it. ___ 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] iLBC codec and mute issues
Hmm. Well, I'm not sure what to try, but it works fine if I use GSM or G711 or even G729. Such a shame, because I liked iLBC. -- Dana On Tue, 22 Mar 2005 22:16:17 +0100, Roman Zhovtulya <[EMAIL PROTECTED]> wrote: > I'm using the mute switch on the Plantronics headset 90 with SJPhone, so > never had this issue :-) > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Dana Olson > > Sent: Montag, 21. März 2005 17:53 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] iLBC codec and mute issues > > > > > > I tried using the iLBC codec, and whlie I like it, I ran into > > a strange issue. I did a few searches on Google and haven't > > found anyone with the same issue as this. > > > > Anyhow, I was using a Plantronics analog headset and box > > plugged into a Digium TDM card, dialed out over my VoIP > > provider's IAX channel to the PSTN. > > > > I was in a conference call which is running on an Avaya PBX > > (which shouldn't matter), and so I muted myself with the mute > > button on the headset box. After a minute or two, I was asked > > to speak again, and so I unmuted, but no one could hear me. I > > tried hitting mute a bunch more times, but still nothing. It > > was making a difference in the headset though; I could hear > > myself a little bit when unmuted, but not when I was muted, > > leading me to believe it was something with the Asterisk box. > > I switched codecs and the issue disappeared. > > > > Does anyone know what the problem is there, seemingly it is > > iLBC, but I was wondering if that's a common thing or not. Is > > this codec unstable like this? > > > > Thanks, > > Dana ___ 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] Reproducible echo on IAX calls to -some- destinations.
On Tue, 22 Mar 2005 12:49:06 -0500, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote: > I'm very, very confused. Dialing out, through VoicePulse, with both gsm > and ulaw CODECs, most of my calls are great. However, calling my > (non-Asterisk) voicemail at my job, and calling my cell phone both > produce horrendous (~ 1/3-second delay) echo. I've tried with different > phones (Polycom and Grandstream), different IAX CODECs (as described, > above), different network connections... but those two destinations are > always horrible, and all my other victim..., errr, test numbers work > just fine, all the time. > > Any idea where to even start troubleshooting? > > Thanks, > > -Ken I've got no solution for you, but I'll be monitoring this thread in case any ideas to try on my end come up, as I have the same issue with my VoIP provider (hi there, I know you're on this list). It's only on occasional calls, but I have to hang up, I can't deal with it. -- Dana ___ 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] why even use SIP
Yeah, if we're making a list, add DNS name resolution to that list. :) -- Dana On Tue, 22 Mar 2005 23:48:46 +0400, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote: > > >If the IAXy had a bit more work done on it, it could be a good option, > >but it's not at the current time. > > > > > Yep! Things like: > > - more codecs (just ulaw? come on...) > - proper DHCP and possibility of static IP > - a 'reset' button > > To start with would be nice to have. > > And my IAXy doesn't work with my european phone (no tone) it's kind of a > drag :( > ___ 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 keep Asterisk up to date on many servers
On Mon, 21 Mar 2005 21:36:53 -0800, Geoff Nordli <[EMAIL PROTECTED]> wrote: > Hi Everyone. > > Asterisk is one of those applications that need to be built from cvs on a > regular basis to keep up with the changes. I have always used package > management tools like apt. > > How does everyone manage their Asterisk servers? > > Geoff I use Asterisk stable, which is updated frequently enough for my liking. I use Debian Woody on my servers. I have Asterisk already running, and my service is interrupted only briefly. Here's what I do: Download the packages to /usr/src/. tar zxvf zaptel-1.0.7.tar.gz cd zaptel-1.0.7/ make clean checkinstall -D --pkgname=zaptel --pkggroup=Asterisk [EMAIL PROTECTED] --dpkgflags=--force-overwrite --nodoc --default -bk --pkgrelease=`uname -r` --pkgversion=1.0.7 mv zaptel_1.0.7-`uname -r`_i386.deb ../debs/ cd .. tar zxvf libpri-1.0.7.tar.gz cd libpri-1.0.7/ make clean checkinstall -D --pkgname=libpri --pkggroup=Asterisk [EMAIL PROTECTED] --dpkgflags=--force-overwrite --nodoc --default -bk --pkgversion=1.0.7 --pkgrelease=1 mv libpri_1.0.7-1_i386.deb ../debs/ cd .. tar zxvf asterisk-1.0.7.tar.gz cd asterisk-1.0.7/ make clean checkinstall -D --pkgname=asterisk --pkggroup=Asterisk [EMAIL PROTECTED] --dpkgflags=--force-overwrite --nodoc --default -bk --pkgversion=1.0.7 --pkgrelease=1 mv asterisk_1.0.7-1_i386.deb ../debs/ cd .. tar zxvf asterisk-sounds-1.0.7.tar.gz cd asterisk-sounds-1.0.7/ checkinstall -D --pkgname=asterisk-sounds --pkggroup=Asterisk [EMAIL PROTECTED] --dpkgflags=--force-overwrite --nodoc --default -bk --pkgversion=1.0.7 --pkgrelease=1 mv asterisk-sounds_1.0.7-1_i386.deb ../debs/ This builds and installs all of the packages I use and then puts the resulting .deb packages into /usr/src/debs. I can then copy the .debs to other servers and just install them with dpkg -i --force-overwrite. Works like a charm for me. -- Dana ___ 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] I need use sip
Heh. It was really funny. If I hadn't seen the preview of the email in Gmail, I probably would've missed this thread entirely. Thanks to you all and Gmail preview for making my day! -- D On Mon, 21 Mar 2005 18:40:42 -0800, Luki <[EMAIL PROTECTED]> wrote: > Nice, gUys... > > Don't pick on Raphael too much. It's an honest mistake. I remember > making some of those -- although not of this caliber -- when I first > learned English a few years back. You really feel stupid once you > realize it... and I don't think the hugs had an implied meaning > either... just a figure of speech in Brazil, I guess. So be nice, > gUys! OK? Nevertheless, it was a good laugh :-). > > --Luki > > Wiley Siler <[EMAIL PROTECTED]>: > > What a great way to end the day! This one has me laughing my ass off > > I am hoping you actually meant "guys". You may want to look up > > the meaning of the word you used... > > ... ___ 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] why even use SIP
On Tue, 22 Mar 2005 12:10:17 +0400, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote: > On a LAN where NAT is not an issue I would go for SIP + decent > hardphones with good echo cancellation. > > On the internet with all sort of NATs + Firewalls, IAX is a must but > unfortunately I don't know of any good, readily available hardphones. If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. -- Dana ___ 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] Can't hear the caller
If your IP address changes daily, you need to continually update this, correct? I too have this issue with SIP, but my ISP is a little unstable, so I need to use DynDNS or No-IP to kinda work around/with it. I assume you can't specify a hostname there? -- Dana On Mon, 21 Mar 2005 13:45:44 -0700, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > i had a similar problem a while ago. I solved it by defining > externip=xxx.xxx.xxx.xxx in sip.conf. It tells the remote SIP client where > you are. > > -chuks. > > > Original Message > Subject: [Asterisk-Users] Can't hear the caller > From: Lane <[EMAIL PROTECTED]> > Date: Mon, March 21, 2005 11:53 am > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > Hi, > > I've got a strange issue, that I haven't found addressed on the wiki. > > My asterisk box is behind a firewall which routes udp/tcp requests on 5060 > and > 8000 to asterisk. > > When I make a call from a Zap or SIP extension on the inside of the firewall > to any Zap or SIP extension on the inside of the firewall, everything works > find. I have access to voipjet, and when I place a call through them > everything also works fine. > > However, I have several SIP extensions defined for remote users. They can > authenticate to asterisk and they can call me and I can call them. However, > whether I place the call or receive the call I cannot hear the remote user!! > They can hear me, though. (I'm inside the firewall). > > My remote users are on XLite, if that makes a difference. > > Anybody got an idea why the firewall is blocking traffic for these SIP > phones, > but not for voipjet? > > Thanks > > lane ___ 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] why even use SIP
On Mon, 21 Mar 2005 10:01:04 -0800, Sys Admin <[EMAIL PROTECTED]> wrote: > I am setting up a new asterisk based call center. I just read: > http://www.voip-info.org/wiki-IAX+versus+SIP > > After reading this and other google results for "IAX vs SIP" is there > any reason why i should use SIP anywhere !! > > t Do you have your voip hardphones picked out yet? ___ 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] Why is asterisk's voice mail called comedian.
On Mon, 21 Mar 2005 08:27:17 -0500, Steve Clark <[EMAIL PROTECTED]> wrote: > It doesn't sound very professional. > > comedian > n 1: a professional performer What's not professional about 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] iLBC codec and mute issues
I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which shouldn't matter), and so I muted myself with the mute button on the headset box. After a minute or two, I was asked to speak again, and so I unmuted, but no one could hear me. I tried hitting mute a bunch more times, but still nothing. It was making a difference in the headset though; I could hear myself a little bit when unmuted, but not when I was muted, leading me to believe it was something with the Asterisk box. I switched codecs and the issue disappeared. Does anyone know what the problem is there, seemingly it is iLBC, but I was wondering if that's a common thing or not. Is this codec unstable like this? Thanks, Dana ___ 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] reply a post
Click on Reply in your mail client, type message, click Send. On Fri, 18 Mar 2005 16:15:24 -, Kanishka Somaratne <[EMAIL PROTECTED]> wrote: > Hi > how do i reply a question asked in this mailling list. > > tks > Kanishka > > ___ > 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: Re[2]: [Asterisk-Users] TDM400P install problems
Can you run dmesg after that command and tell us what the relevant output is? On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi <[EMAIL PROTECTED]> wrote: > Hello Dana, > > Friday, March 18, 2005, 3:23:36 PM, you wrote: > > DO> If you have any FXS ports, use wcfxs. > > No, only green modules. > > But this is what I get when loading driver > > modprobe wcfxs > FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): > Unknown symbol in module, or unknown parameter (see dmesg) > FATAL: Error running install command for wctdm > > What relates wcfxs to the wctdm that I was using previously ? > > Maybe deleting wctdm > > DO> On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi > DO> <[EMAIL PROTECTED]> wrote: > >> Hi, > >> > >> I was using a TDM400P with cvs version of asterisk, loading the driver > >> with "modprobe wctdm". > >> > >> Some days ago I switched to stable version 1.0.6, where I found no > >> trace of such module ... is wcfxo to be used instead ? > >> > >> Do I also have to change something in zaptel.conf ? > >> > >> Tnx for any help! > >> > >> -- > >> Best regards, > >> Alessio mailto:[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 > >> > DO> ___ > DO> Asterisk-Users mailing list > DO> Asterisk-Users@lists.digium.com > DO> http://lists.digium.com/mailman/listinfo/asterisk-users > DO> To UNSUBSCRIBE or update options visit: > DO>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Best regards, > Alessiomailto:[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
Re: [Asterisk-Users] TDM400P install problems
If you have any FXS ports, use wcfxs. On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi <[EMAIL PROTECTED]> wrote: > Hi, > > I was using a TDM400P with cvs version of asterisk, loading the driver > with "modprobe wctdm". > > Some days ago I switched to stable version 1.0.6, where I found no > trace of such module ... is wcfxo to be used instead ? > > Do I also have to change something in zaptel.conf ? > > Tnx for any help! > > -- > Best regards, > Alessio mailto:[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] Dial multiple extensions, but different variables/timeouts
On Wed, 16 Mar 2005 19:58:55 -0800, Luki <[EMAIL PROTECTED]> wrote: > Anyway, if anyone ever needs this info, they can Google it now :-). Might be a good thing for the wiki too. ;) ___ 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] Best Grandstream firmware to use?
I have that on my HandyTone 286, and it's got some issues. At least mine does. The page loading in FireFox very rarely ever completes, and this means that I can't really provision this thing very well at home (I only run Linux). I brought it to work and it loads up fine in IE here. I engaged their tech support and their initial response was "did you refresh the page?" ... Anyhow. -- Dana On Thu, 17 Mar 2005 07:48:11 +1100, Rod Bacon <[EMAIL PROTECTED]> wrote: > I use 1.0.5.22 > > Can't fault it. > > Don't be afraid of upgrading to a newer version, you can always downgrade > again. > > - Original Message - > From: "Paul Fielding" <[EMAIL PROTECTED]> > Sent: Tuesday, January 18, 2005 2:34 AM > Subject: [Asterisk-Users] Best Grandstream firmware to use? > > > I've seen lots of stuff go around about Grandstream firmware levels (in my > > case specifically the BT101/102). I'm just wondering what the currently > > accepted 'best' firmware version is to use? After seeing stuff going > > around about buggy firmware I want to know what I'm getting into before > > upping past my current 1.0.5.11.It's relatively stable, and the last > > thing I want to do is update to a flaky firmware > > > > Paul > > -- next part -- > > An HTML attachment was scrubbed... > > URL: > > http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/1d3c83e1/attachment.htm > > > > ___ > 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] Digium : no lead time!
I emailed ipVolution about it, and they don't have it out yet. Their cards should be coming out soon, they say. Although the target release month was last month, I believe. -- Dana On Wed, 09 Mar 2005 15:21:35 -0800, Chris A. Icide <[EMAIL PROTECTED]> wrote: > If you do a web search for ipVolution TDM120, you should find someone who > claims to have a card that does such a thing. > > -Chris > > > On 02:21 PM 3/9/2005, Brandon Patterson wrote: > > > >Supposed to be someone in Minn. working on this. I heard the name Dan. > >He might be in the hardware biz. > > > > > >> On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote: > >>> Again, may be off topic but are there any cards out there supported by > >>> asterisk that have on-board DSPs to do better 729->711 or 729->PRI > >>> conversion? > >> > >> Not yet, and I don't know if anyone is working on the drivers for such a > >> card. > >> -- > >> 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 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] Hardphone deployment recommendation
Thanks for your replies. My main concern is to keep the price down. If the BudgeTones are crap phones, which previous posts to this mailing list seem to indicate, and we have to replace them often, then the price for a better phone would be worth it. I don't think we need 3-way calling either, and as stated already, nothing a proper dialplan can't fix. I'm going to try getting in a couple of those Sipura 841s for testing. Thanks for that suggestion. -- Dana ___ 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 Codec
I've called using G729 SIP phones over my LAN, and I think it sounds quite good. YMMV. On Fri, 04 Mar 2005 15:58:23 -0500, Martin Roy <[EMAIL PROTECTED]> wrote: > I have 2 Asterisk servers connected with IAX. It's working fine I can > call an extension from one phone in an office to another phone in the > other office. The only problem I have is lagging. What codec should I > use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I > configured it to disallow all and use GSM only. In my sip config of each > phone I use disallow all and allow ulaw and alaw only. > > I see that the Cisco phone support G.729a, G.711u and G.711a codecs. > > I know that Digium is selling G.729 codec on there website. Should I get > that to fix my problem? > > The less bandwitdh the codec takes over the Internet the better chance I > have that it will work fine I presume. > > The maximum upload speed I can reach is 64k/sec so 512kbits. But that's > if there's nothing else using the internet. I see that G.729 use 8kbps > per call which seems quite small. The sound quality is good enough for > SIP to SIP phones over the Net? > > Thanks > > Martin > ___ > 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] Hardphone deployment recommendation
I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some people seem to recommend the Sipura adapters. I'm looking to keep my cost down, and the BudgeTone is around $100 CDN, give or take. Let me know what you would purchase for about 100 users, 1 line each, G729, and why. We've had decent results with the BudgeTone phones I have already, but I only have about 4 of them, and I have about 5 Aastra/Sayson 480i phones which are a bit pricey for this application, and very featureful. The Sipura box I have is alright, and the IAXys work, but aren't an option for this application. I'm looking for SIP, not IAX. The main reason I ask to the mailing list instead of basing a large purchase decision on the phones I have here is that while these devices haven't failed on me yet (with the exception of one flaky 480i), I know that there are some of you who have experience with large deployments. Also, if you recommend an analog adapter, is there any recommendation for analog phone to go with it? I'm not sure if the users will want headsets or handsets, so either one is fine. Thanks for any advice and experiences. PS: If you're thinking you'll get a purchase contract out of me, you won't - the supplier decision isn't in my hands, so don't bother spamming me with your deals. -- Dana ___ 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] Any experience with Sangoma cards?
Have you made any progress on this? -- Dana On Mon, 24 Jan 2005 11:38:52 -0500, Jon Bebeau <[EMAIL PROTECTED]> wrote: > I'm exactly in the middle of benchmarking the A104 and T410p. I'm > developing a matrix of CPU, bandwidth throughput and trying to find high > water marks under several loads; single processor, multi processor, Xeon vs. > P4, Hyperthreading vs. not, and mixed voice and data T1s for the Sangoma > boards. It's turning out to be an ordeal. I'll post the findings when I > results. > > Jon > > - Original Message - > From: "Robert Augustyn" <[EMAIL PROTECTED]> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Sent: Monday, January 24, 2005 10:22 AM > Subject: RE: [Asterisk-Users] Any experience with Sangoma cards? > > > Jon, > > Would you care to comment on how have you been using these? > > Thanks > > robert > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Jon Bebeau > > Sent: Monday, January 24, 2005 8:52 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Any experience with Sangoma cards? > > > > Hello, > > > > Yes. I've had good experience with all three you mentioned. > > > > Jon > > - Original Message - > > From: "Robert Augustyn" <[EMAIL PROTECTED]> > > To: > > Sent: Sunday, January 23, 2005 2:17 PM > > Subject: [Asterisk-Users] Any experience with Sangoma cards? > > > > > >> Hi, > >> I am considering A101/102/104 cards for my asterisk installations. > >> Has any of you used these or any Sangoma cards in such environment? > >> Any thoughts? > >> How do they stack up against Digium cards? > >> Any input would be greatly appreciated. > >> 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] X-Lite Softphone
I've had the best luck with SJphone. Using a USB headset really helps as well, opposed to the sound card or onboard audio with a standard headset. I don't like the skins that come with most softphones, and XLite is no exception. SJphone lets me disable it, and the profiles are nice, since you can copy them from computer to computer. I haven't tried XPro or eyeBeam. -- Dana On Tue, 15 Feb 2005 08:56:48 -0800, Richard J. Sears <[EMAIL PROTECTED]> wrote: > Hey Everyone, > > I downloaded and installed the X-Lite softphone the other day (the lite > version) and cannot seem to get it to work well. > > Don't get me wrong, it registers with my asterisk server and everything > seems to work well, except the call quality really is horrible. > > I thought it may be the place I was trying it at (DSL) so I took it to > the office and tried it right next to the asterisk box and had the same > luck. > > My laptop is the Dell XPS, so power, ram, etc are not problems, and > loading it onto my desktop system revealed the same results. > > There was also no difference between a NAT implementation and a regular > (live IP) implementation of the software. > > I am getting stuttering speech, cutouts, etc all the time. > > Running my Cisco 7960 at the same locations and it works fantastic with > no issues at all. > > Is anyone else using this softphone or does anyone know of a better > softphone or some hints on configuration that may make X-Lite work > better..? > > TIA > > ** > Richard J. Sears > Vice President > American Internet Services > > [EMAIL PROTECTED] > http://www.adnc.com > > 858.576.4272 - Phone > 858.427.2401 - Fax > INOC-DBA - 6130 > > > I fly because it releases my mind > from the tyranny of petty things . . > > "Work like you don't need the money, love like you've > never been hurt and dance like you do when nobody's > watching." > > ___ > 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] ATA's
There are tons of comments about the S100i, or IAXy, as it's called. Can't say I like it myself, we have 3 of them right now, and they get really hot and for some reason don't have the MAC address labeled on them, and also we couldn't get them to actually take an IP from a Microsoft DHCP server (worked fine with Linux/Linksys). Also, the IAXy provisioning is annoying. But this is just my thoughts based on my experience. -- Dana On Tue, 15 Feb 2005 12:29:42 -0500, Erick Perez <[EMAIL PROTECTED]> wrote: > what about the digium S100i, haven't used but any comments? > i know it's only one fxs one lan port does g711 also. no g729. > > > On Tue, 15 Feb 2005 10:29:35 -0500, Mark Eissler <[EMAIL PROTECTED]> wrote: > > > > On Feb 15, 2005, at 3:17 AM, Voip Business wrote: > > > > > hello, my experience > > > > > > 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE > > > 2.- MTA-V102 > > > 3.- Sipura spa 2000 > > > 4.- Granstream > > > > > > > > > ATA186 SUXs > > > > I can't speak so fondly of the Azatel which I had sitting around after > > a canceling a VOIP service. Maybe I just need a new firmware rev (but > > they don't exactly make those available at the Azatel site). Plus, the > > web interface is excruciatingly limited. I mean, you can't even > > configure echo cancellation. > > > > I think the ATA186-L2 is kind of pointless at this stage. It's old > > hardware...although Cisco did end up issuing a firmware update last > > year. Still, there's got to be some reason why Cisco as switched to > > using a Sipura produce (the PAP2)BTW the ATA186 was designed by > > some of the Sipura folks as well. > > > > My choice is still Sipura-branded equipment. There's no way of knowing > > how often firmware will be released for the Linksys-branded stuff or > > what level of support there will be. > > > > -mark > > > > -- > > Mark Eissler, [EMAIL PROTECTED] > > Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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 > > > > -- > > --- > Erick Perez > Linux User 376588 > http://counter.li.org/ (Get counted!!!) > Panama, Republic of Panama > ___ > 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] Linphone / Kphone
On Mon, 14 Feb 2005 20:01:18 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote: > Another point to note is that seemingly all closed source softphones > (SJ, XLite beta and also cornfed) make connections to web servers > and transmit platform/call information. Don't know how you think about > that, but for me that's behavior I'd like to avoid if ever possible. > > Regards, Bruno. Do you have this documented somewhere? Is this for the Linux Xlite and SJphone only, or the Win32 ones as well? -- Dana ___ 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] Uptime/reliability with SER, Asterisk
I really appreciate your reply. For Asterisk, are you using G729 as your codec, or something more high-bandwidth (ulaw)? Is there any definition of stable that you would use that would point to SER and Asterisk not being "stable"? Again, thanks for your reply. -- Dana On Mon, 14 Feb 2005 13:27:53 -0500, Steve Blair <[EMAIL PROTECTED]> wrote: > > Our SER/Asterisk implementation is extremely stable if you define > stable as the ability to deliver a set of features without either > application > crashing. We are a production environment with 75 users total. Asterisk is > only used for voicemail. The only issue we have is that the audio > (greeting or message) being play from Asterisk sometimes has a > robotic or "stuttering" quality to it. I suspect this is latency in the > data network but I have yet to figure it out. > > -Steve > > Dana Olson wrote: > > >Could anyone shed any light on how SER and/or Asterisk (stable branch) > >has held up for them in that last while? > > > >Are you using SER and/or * in a production environment? Do you ever > >restart the software or reboot the system? How many users are > >utilizing the system? How many calls per day/concurrently? > > > >I read some uptimes and such on the mailing list from long ago, so I > >was wondering what some more recent results were like. I'm running > >Asterisk at home, but only since recently so my experience won't be a > >good representation of the reliability and stability. > > > >Thanks in advance. > >___ > >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 > > > > > > -- > > ISC Network Engineering > The University of Pennsylvania > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > > voice: 215-573-8396 > > 215-746-8001 > > fax: 215-898-9348 > > sip:[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] Uptime/reliability with SER, Asterisk
Could anyone shed any light on how SER and/or Asterisk (stable branch) has held up for them in that last while? Are you using SER and/or * in a production environment? Do you ever restart the software or reboot the system? How many users are utilizing the system? How many calls per day/concurrently? I read some uptimes and such on the mailing list from long ago, so I was wondering what some more recent results were like. I'm running Asterisk at home, but only since recently so my experience won't be a good representation of the reliability and stability. Thanks in advance. ___ 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 scary log (OT)
On Fri, 11 Feb 2005 09:00:49 -0500, Dana Olson <[EMAIL PROTECTED]> wrote: > On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> > wrote: > > Derek Whitten wrote: > > > I also call bullshit.. OpenBSD does NOT allow ssh root login by > > > default.. why do you think that they have such an excellent security > > > track record.. > > > > Derek, > > > >I am sorry to say, that in fact, OpenBSD does allow SSH root logins > > by > > default: > > > > http://www.openbsd.org/cgi-bin/cvsweb/~checkout~/src/usr.bin/ssh/sshd_config?rev=1.70&content-type=text/plain > > > > BTW, OpenBSD's track record of security has nothing to do with whether > > they allow root logins by default or not. If the admin isn't wise > > enough to pick a -decent- root password they shouldn't be running a box > > connected to the internet. > > > > I think people need to start to provide HARD FACTS in some of these posts. > > > > I don't see what any of this has to do with Asterisk... > > > > -- > > Kristian Kielhofner > > Unless I'm missing something, the only line that is ENABLED in that > file is this one: > > Subsystem sftp/usr/libexec/sftp-server > > The rest appear to be commented out with #, unless I'm not > understanding how that all works... > -- > Dana > Nevermind. I see how it is... Good thing I'm not a BSD admin. -- Dana ___ 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 scary log (OT)
On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: > Derek Whitten wrote: > > I also call bullshit.. OpenBSD does NOT allow ssh root login by > > default.. why do you think that they have such an excellent security > > track record.. > > Derek, > >I am sorry to say, that in fact, OpenBSD does allow SSH root logins by > default: > > http://www.openbsd.org/cgi-bin/cvsweb/~checkout~/src/usr.bin/ssh/sshd_config?rev=1.70&content-type=text/plain > > BTW, OpenBSD's track record of security has nothing to do with whether > they allow root logins by default or not. If the admin isn't wise > enough to pick a -decent- root password they shouldn't be running a box > connected to the internet. > > I think people need to start to provide HARD FACTS in some of these posts. > > I don't see what any of this has to do with Asterisk... > > -- > Kristian Kielhofner Unless I'm missing something, the only line that is ENABLED in that file is this one: Subsystem sftp/usr/libexec/sftp-server The rest appear to be commented out with #, unless I'm not understanding how that all works... -- Dana ___ 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 scary log
On Thu, 10 Feb 2005 17:49:23 +, Clive Carter <[EMAIL PROTECTED]> wrote: > I hesitated before sending this, as I have been flamed before for being a > beginner. but > I am newish to linux/asterisk, and I am running an ssh server. It is still > running with default settings, (I dont know yet how/where to change it), and > I CAN logon remotely as root. > (Haven't figured out how to 'su' yet !) > > This is using the Rapid Xorcomm v 1.0 cd, which I believe (may be wrong) is > based on a very recent version of Debian ? > Perhaps xorcom have changed the default setting ? > > -- > Clive > > Email : [EMAIL PROTECTED] > Tel : 08444844790 >Alt : 08450043366 > Fax : 08444844813 > SIP : [EMAIL PROTECTED] > Mobile : 07031945504 Hey Clive. I thought it was mentioned earlier before in the thread, but if not, all you need to do is edit your sshd_config file. In Debian, this is located at /etc/ssh/sshd_config, but it could be different for other distros. Open that up in a text editor and then locate the line that says PermitRootLogin yes, and change that to PermitRootLogin no. Save it, and then restart SSH. On Debian, you type in /etc/init.d/ssh restart, but on other distros it might be different. Note that you'll have to be root to edit that file and restart that service. -- Dana ___ 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 scary log
On Thu, 10 Feb 2005 11:44:37 -0600, Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Thu, 2005-02-10 at 11:36 -0500, Noah Miller wrote: > > > IMO, your best defence is leaving ssh's default setting > > > which disallows root logins entirely. There's no reason > > > for a remote user to ever have to log in as root. Root > > > access should be obtained by a logged-in normal user > > > using sudo, or su. > > > > I'm not sure what happens when you do a fresh compile and > > install of OpenSSH, but every distro I've ever worked with > > (Red Hat, Gentoo, Slackware, Vector, Tao, Yellow Dog, > > Debian, Knoppix, SuSe, Linspire, FreeBSD, OpenBSD, Darwin, > > OS X) has allowed root logins via SSH by default. Maybe > > they're changing that on newer versions of some distros. > > I dunno. > > I'll call bullshit on that. I know for a fact that Debian does NOT allow > root logins except from console. Hell Debian isn't allowing root logins > from X anymore due to the likely hood for you to try and use root for > more than administration. > > I know Mandrake does annoying things if you try to login as root on > anything but console to also discourage it's use. > > I don't expect much from Linspire as it attempts to be windows. As for > the rest in your list other than OS X, I wouldn't bother trying to run > them when you have Debian available. > -- > Steven Critchfield <[EMAIL PROTECTED]> Testing and Unstable might not, but Stable does. I've had to change it in the last 3 Woody installations I've done in the last three weeks. I'm talking about SSH specifically. -- Dana ___ 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] G.729 codec for X-lite soft phone
And for Windows, a minimum purchase of 1 unit... Is he using Mac or PocketPC? If not, then he doesn't have to worry. On Wed, 09 Feb 2005 15:59:45 -0500, Sergey Kuznetsov <[EMAIL PROTECTED]> wrote: > Adrian, > > Have you ever read the note for that? > > =head Comment > > * NOTE > G.729a for X-PRO Pocket PC is available with a minimum order > of 20,000 units, > G.729a for X-PRO Mac OSX is available with a minimum order > of 10,000 units, > G.729a for LindowsOS is not available at this time. > =cut > > > Adrian Chapman wrote: > Daniel Eboa wrote: > Sir, > > I think when somebody asked a question, is because he doesn't know the > > answer. Even maybe when for some people like you, the answer is > > evidence. > > Thinking that I know the answer of the question I asked, suppose that > > I'm stupid, while I'm not. > > I you feel offence by the question I asked, please simply ignore it. > > Regards. > > Daniel. > > Daniel, > > I think you'll find that Seshu was implying you do a little basic research > for yourself. It's easier than asking here, and you get an answer more > quickly. > > A two second visit to www.xten.com, and clicking on the clearly labelled > link to compare the free and paid versions of the softphone shows that one > difference between the free and paid versions is that the paid version > supports G.729a. Get your credit card out, and your desires are met. > > http://www.xten.com/index.php?menu=products&smenu=compare > > > > -- All the Best! Sergey. = Sergey > Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: > http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business > phone: (416) 548-9700 Mobile phone: (647) 287-8448 > ___ > 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] Re: Analogue Line to Asterisk (Which Digium Model???)
On Wed, 9 Feb 2005 14:59:44 -0500, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Walid - > > > I need to use Asterisk to call out PSTN numbers via an analogue line. I > > understand Digium manufactures these kinds of cards, but can someone > > tell me > > which model number it is. I really only need a card with one or 2 > > analogue > > ports max. > > You'd be looking for the TDM400P. You can get it in various > configurations of FXO ports (to the phone company) and FXS ports (to > your phone handsets). You might want the TDM11B (one FXS and one FXO), > or the TDM02B (2 FXO's), or just the TDM01B (one FXO). > > - Noah Yeah, and a good thing about the TDM card is, say you get the TDM01B and then later on down the road, you want to add another FXO or FXS, you can buy just the module and add it to that card. You can have up to four in total. I was considering getting a X100P, but ManxPower (that's his alias on IRC, I think his name is Eric) recommended the TDM card and I totally don't regret it. -- Dana ___ 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] Startup Question
On Wed, 9 Feb 2005 13:46:41 -0600, Anton Krall <[EMAIL PROTECTED]> wrote: > Guys, Im new to asterisk and voip but Im have a couple of questions > regarding the initial setup. > > 1. Im going to install an asterisk server at home, where I have 2 phone > lines, what kind of card do I need to get? I was thinking about 2 X100P > Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? > Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS > devide for attaching analog phones right? Any phone? Is it better to use > those or use IP Phones? > > The software solution for voip like sokol's one seems a bit unstable due to > winxp limitations :) so Im thniking about going the HW way.. What do you > think? Are those IAX compatible? > > 2. I checked stuff about IAXNet and FWD, but Im still unclear as to what > really FWD is? And how does it help you on? For example, Im in Mexico City, > can I setup my asterisk box and for example, how can I use it so that I can > make calls to the US by using a US dialtone? Of what does FWD really does? > Is it a network of asterisk boxes connected to each other so FWD users can > talk to each other? Or is it a network with POTS line that can help you get > international dialouts? > > Thank you for your patience as I can imagine this has been asked a lot of > times :( > > __ > Anton Krall I'd recommend a TDM22B card. That would give you 2 FXO and 2 FXS all on one card. I'm using a TDM11B, with, you guessed it, only 1 FXO and 1 FXS. It works great in my old Pentium 2 system. I have a cordless phone plugged into it, and it works great. I've also got a HandyTone 286 on the way, for plugging up another phone downstairs. I have a Packet8 DTA that is hackable to some extent and *works* with a phone plugged into that, but I don't recommend wasting your money. Can't help you with FWD, as I haven't touched it yet. I've signed up with a local IAX provider, but only have had a few days' experience. I canceled my land-line's long distance today. :) -- Dana ___ 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] Web based Asterisk management tool
Yeah, I wouldn't mind having a look at it as well. -- Dana On Wed, 9 Feb 2005 09:36:27 -0800, Michael Levenson <[EMAIL PROTECTED]> wrote: > Why not share with the community? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler > Sent: Wednesday, February 09, 2005 9:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Web based Asterisk management tool > > Gary, > contact me off-list. I have developed a GUI Windows based tool that will > allow > management of configuration files if you are running RealTime. It supports > sip,iax,extensions,voicemail currently. It will also display CDR's and the > various > schema's used by the Asterisk box. > > Tom Chandler > [EMAIL PROTECTED] > - Original Message - > From: "dean collins" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, February 09, 2005 10:47 AM > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > > You need to go back and reread. > > > > It is just pretty much an asterisk configuration tool (ok some minor > things in the backend but it's the best out there). > > > > AMP is available for free download but they make their money by offering > support. > > > > [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web > Meetme. > > > > If you really have a need to support thousands of extensions as you > suggest then you should really go back and learn how to program asterisk > with a database yourself from scratch. > > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary > > Sent: Wednesday, February 09, 2005 11:17 AM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > > > Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured > distribution > > for small systems?? I am looking for an open source web management tool to > > use on any size asterisk server (even ones that are already up and > running) > > the user base could be anything between small and large with many external > > lines, > > > > Ive looked at AMP, is it free ? and are there any alternatives or is AMP > the > > only open source web management tool ? > > > > -Original Message- > > From: dean collins [mailto:[EMAIL PROTECTED] > > Sent: 09 February 2005 15:05 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > > > That would be the AMP database, I don't know. > > > > Ping the amp list and find out. > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa > > Sent: Wednesday, February 09, 2005 9:47 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > > > How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of > > users > ? > > > > Regards. > > > > Daniel. > > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of dean collins > > Sent: mercredi 9 février 2005 15:42 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > > > Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download > > at sourceforge and does exactly what you are looking for. > > > > > > Cheers, > > Dean > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Brett, > > Gary > > Sent: Wednesday, February 09, 2005 8:01 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Web based Asterisk management tool > > > > > > Hi there > > > > I am new to Asterisk and am looking for a web based management tool, for > > managers to manage hunt groups, extensions etc and for user to have > > access > > to there own phone features. I have seen there are a number of > > commercial > > tools available for this, but I presume there are some freeware options > > too > > > > I noticed one that I like at http://www.thirdlane.com/screenshots.htm > > but I > > am assuming this is just a freeware product that has been re-badged so > > to > > speak. > > > > If any body can give me some suggestions that would be great > > > > Regards > > Gary ___ 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 and SER Integration together
On Wed, 9 Feb 2005 11:44:30 -0500, Paul Rodan <[EMAIL PROTECTED]> wrote: > > > I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive > benefits, however, my initial playing around in SER's configuration > indicates it's NOTHING like Asterisk at all, and almost 5x as difficult to > understand and configure. But that's only after a few hours of playing with > it. > > > > I'm interested in learning SER more, especially the integration with > Asterisk. Is there a good how-to guide with lots of examples on how to > accomplish this optimal setup? Anybody got any good links or resources or > can help me with examples? Right now I have Asterisk doing all the work and > it's getting frustrating w/ Quality issues left and right and such. It's not nearly as easy as Asterisk is to understand, but it works. I have really very limited experience with it, but just to give you an example of the support out there for it: I was trying to figure out how to forward a call from a user who called to [EMAIL PROTECTED] to [EMAIL PROTECTED] and I tried many different ways. I couldn't get it right. I read the how-to, read through the document, and it wasn't very clear. I went onto IRC and asked for help, and someone told me that I should just use Asterisk. I said I didn't want to, and gave reasons why. They said that to do a forward it was very complex, and I'd be better off just using Asterisk or hiring a consultant to build it for me, and that they had no idea how to do it. Anyhow, I figured it out finally, after reading a totally unrelated mailing list thread. Fast forward a half hour, this same person asks me how many lines we have at my company and states that they would like some of our business... SER seems powerful and it's all the rage, and from what I can tell, it'll be fun to learn it... I'd like to see better documentation for it, and if I ever get to the point where I can provide it, I will. ___ 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] problem with running ztcfg
On Wed, 9 Feb 2005 08:23:57 -0800 (PST), Paul Chan <[EMAIL PROTECTED]> wrote: > Hi All, > > I just installed Asterisk 1.0.5, and the > installation went fine (I ran modprobe zaptel and > modprobe wcfxo). However, when I ran ztcfg I get the > following: > > ioctl(ZT_LOADZONE) failed: Invalid argument > Notice: Configuration file is /etc/zaptel.conf > line 135: Unable to register tone zone 'us' > > After that I ran Asterisk and it seem to started ok, > except that it won't pick up any calls. Has anyone > seen this or know what could be the problem? > > Thanks for your help in advance! What hardware are you using? What is your zaptel.conf and zapata.conf? ___ 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: Digium TDM400p troubles
On Tue, 8 Feb 2005 12:30:36 -0500, Noah Miller <[EMAIL PROTECTED]> wrote: > > I installed a tdm400p into a old p2 machine. > > I'm not able to see it under /proc/interupts or using lspci.. > > we removed all other cards. changed slots, forced irq to that slot.. > > etc etc. > > > > what is the min specs needed to get one of these cards running? > > PCI 2.2 > > I don't think the TDM cards will work on PCI 2.1 or lower. Check your motherboard for that. I am running a TDM11B successfully on an old Dell MT400 workstation, which is a Pentium II-based system (dual 333MHz CPUs). ___ 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] newbie questions
I run Debian, and it's not hard to get a base install running. If you want a GUI and such, then it'll be more than "follow the screen prompts." I've been writing some Debian documents, if you're interested, email me off-list. Anyhow, on pretty much any distro, you can make your own packages (RPM, DEB, TGZ, whatever) from compiling source using the program called CheckInstall. It'll make removing your programs much easier, and if you deploy many similar systems, you can reuse the packages. I did this when I ran Mandrake, and it worked great, especially when Mandrake didn't have a lot of the software that I wanted to use. More often than not, Debian has what I want, but if I can't find something (ipkungfu, for example) then I'll use checkinstall to make it easy to remove in the future. -- Dana On Tue, 08 Feb 2005 17:06:14 +, Mark Benson <[EMAIL PROTECTED]> wrote: > Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is > what I am using asterisk for. > > I would have thought mandrake would have been ok - but haven't used it > for a while. I'm running FC2 (fedora core2) and asterisk complies and > runs without any problems. > > Dont fear make. Apps, for the most part, compile really easily on linux. > Follow the instructions to the letter and you shouldn't go wrong. Its > often as simple as typing make waiting a bit for stuff to stop happening > and then typing make install. Asterisk prompts you with various other > options like make help and make samples (or something like that) so its > pretty straight forward. > > You don't need any cards for asterisk. No phone cards, no sound card > just whatever allows you to connect to your lan and/or the internet. > > g729 - isn't required. There are plenty of other codecs you can use for > free. > > Accounting, cdr, ser etc - I haven't got that far myself either. > > Shaoul Jacobson - TELLINK wrote: > > >Hi, > > > >I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) > > > > > >1. the distro > >I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing > >(some C or C++ or python ...) > >(buy the full version ) > > > >maybe the latest fedora is more complete ? > >or easier to complete with rpmfind > >(I am green to linux too, but I open my windows & gates to the tux) > > > >(bsd, debian are a bit too tech for me yet, no flaming please.) > >I prefer ready made rpm's or alike than compile AT THIS TIME. > >(I promise to improve over time) > > > > > >2. download > >any rpm ? or I must download sources and 'make install' ? > >(I found one iso, but it seemed to require a pstn card) > >(RTFM a second / third time could is always a good option) > > > >3. pure VoIP > >is it ok to use it in pure VoIP mode without any 'phone cards' ? > >all (most) settings & samples I see include such cards. Needed or not ? > > > > > >4. g729 not free. > >It seems that requires some licensing to digium. > >Can that be without using any 'card' (just VoIP) ? > >How to control the licenses then ? > >(I e-mailed them the question, but got no answer) > > > > > >accounting, cdr's, ... that's for later > >(first I have to be able to phone) > > > > > >regards, > > > >Shaoul Jacobson > >___ > >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] TDM400 Problem
On Mon, 7 Feb 2005 23:16:05 -0600, Eric Rees <[EMAIL PROTECTED]> wrote: > Has anyone seen this message trying to install an TDM400.. spurious > 8259A interrupt: IRQ7 > > This error happens after I do a modprobe wctdm and then the system > hangs. I am installing this in an Asus motherboard with a VIA P4M266 > chipset. I used to get this message a lot on my computer (before I even heard of Asterisk, mind you). When I looked into it and asked people questions, I was told that it is a harmless message. I don't know if this is the case with Digium hardware though... It could cause some issues. Is the card working for you at all? ___ 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] Question about TDM11B Configuration
On Tue, 8 Feb 2005 11:56:18 +0200, Yousri Farouk <[EMAIL PROTECTED]> wrote: > Hello all, > > i would like to configure TDM11B with Asterisk, if any one have the > configuration steps please provide me it. > > Thanks in advance Have you tried looking at Digium's site?? http://www.digium.com/index.php?menu=documentation Try the wiki: voip-info.org ___ 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 on a single phone line
On Mon, 07 Feb 2005 11:08:19 -0800 (PST), Job 317 <[EMAIL PROTECTED]> wrote: > I have asterisk installed on a linux workstation (1 phone in and 1 phone > out jack). > > I want to configure Asterisk to show me any available data about any > calls (i.e. phone numbers, caller-id) as well as screen unwanted calls > to voice mail. > > My first question is, what (if any) specific kernel support do I need? I > build my own kernels and I usually turn off telephony and isdn support. > > Second, which document(s) discuss configuration of Asterisk for this > purpose? > > Thanks, > > Job What hardware do you have to connect your phone and line to your PC? I use a Digium TD11B and it works well. I use custom kernels on Debian too, and I turned off all telephony support. Everything you need for a Digium board and Asterisk aren't in the kernel. -- Dana ___ 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] HP ProLiant server for Asterisk
On Fri, 04 Feb 2005 15:53:13 -0600, Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Fri, 2005-02-04 at 16:02 -0500, Dana Olson wrote: > > > Out of curiosity, do any of you think that the ML350 would be a better > > choice, with similar options? They're a lot cheaper too, and I haven't > > found any negative reports (no solid positive reports either) yet. The > > Debian install will be a bit annoying, but that's alright, as long as > > it's do-able. > > I'm curious as to why you think debian wouldn't be anything but straight > forward to install o that machine. I didn't see any hardware on there > that shouldn't be super easy to get working. Also Debian installers have > come a ways in the last series. I was disappointed to not have at least > a little hiccup on a brand new machine I installed recently. > > -- > Steven Critchfield <[EMAIL PROTECTED]> Hey Steven, I'm talking about Debian Stable, if that makes a difference. The particular disc that I want to standardise on includes the 2.4.18bf24 kernel, and apparently the onboard ethernet is not supported by this kernel. Also, if we're using the 64x SmartArray controller, this is also not supported by the default kernel. I don't believe these options showed up until 2.4.21 or so. This means that I'd have to install to a SCSI drive off of the onboard SCSI first, and then cp the data to the SmartArray after I've upgraded the kernel, and then modify LILO and /etc/fstab to point to the new devices. I've never liked Debian Testing for regular use, and Sid is great on my home desktop and laptop, but even on my own home server I run Woody. -- Dana ___ 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] HP ProLiant server for Asterisk
On Fri, 4 Feb 2005 14:35:14 -0500, Dana Olson <[EMAIL PROTECTED]> wrote: > On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe > <[EMAIL PROTECTED]> wrote: > > On Fri, 4 Feb 2005, Dana Olson wrote: > > > > > I'm looking at ordering a server from HP. I checked around on Google > > > and found in the Wiki that the ProLiant DL380 is supposed to be known > > > to work with *. > > > > > HP ProLiant DL380 G4 Server w/ the following options: > > > Intel Xeon 3.20GHz/1MB > > > 2GB REG PC2-3200 (2 X 1GB) > > > HP ProLiant Battery Backed Write Cache Enabler for SA6i > > > RAID 1 drive set > > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > > > Hot Plug Redundant Power Supply Module > > > HP Redundant Fan Option Kit (3 fans) > > > 1.44MB Floppy Disk Drive > > > Slimline 24X CD-ROM > > > > Nice setup I think. The only thing with the DL380G3 and G4 is that it > > freezes in some situations. This has been reported in the company > > where I work and is a known bug to HP which will be solved by some > > firmware upgrade. > > > > However the problem occurs only in some special situation noone can > > really describe (cosmic rays ?! ;-) and at that location where I work > > we installed SAP on a DL380G3 and it never froze. > > > > And at the end let me say that personally I like the DL380s. You can > > even resize Logical Drives of the RAID (e.g. when putting in > > some new harddisc) without shutting down the server! > > > > Good Luck! > > > > Christoph > > > > Thanks for all of the replies, both on and off list. > > I should have also mentioned that I'll be installing Debian Stable on > it. Are you using Debian or something else? > > There seems to be an issue with the G4 and the Digium TE cards, so I'm > unsure about placing the order at this time. I've been eyeing the > Sangoma A104 card and the ipVolution TDM120 (not out yet, I believe) > as an alternative. > > I have a service contract with an HP provider, so I'd prefer to stick > with an HP server, if at all possible. However, if the T1 interfaces > cause issues, there may be no choice. > -- > Dana Out of curiosity, do any of you think that the ML350 would be a better choice, with similar options? They're a lot cheaper too, and I haven't found any negative reports (no solid positive reports either) yet. The Debian install will be a bit annoying, but that's alright, as long as it's do-able. -- Dana ___ 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] HP ProLiant server for Asterisk
On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe <[EMAIL PROTECTED]> wrote: > On Fri, 4 Feb 2005, Dana Olson wrote: > > > I'm looking at ordering a server from HP. I checked around on Google > > and found in the Wiki that the ProLiant DL380 is supposed to be known > > to work with *. > > > HP ProLiant DL380 G4 Server w/ the following options: > > Intel Xeon 3.20GHz/1MB > > 2GB REG PC2-3200 (2 X 1GB) > > HP ProLiant Battery Backed Write Cache Enabler for SA6i > > RAID 1 drive set > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > > Hot Plug Redundant Power Supply Module > > HP Redundant Fan Option Kit (3 fans) > > 1.44MB Floppy Disk Drive > > Slimline 24X CD-ROM > > Nice setup I think. The only thing with the DL380G3 and G4 is that it > freezes in some situations. This has been reported in the company > where I work and is a known bug to HP which will be solved by some > firmware upgrade. > > However the problem occurs only in some special situation noone can > really describe (cosmic rays ?! ;-) and at that location where I work > we installed SAP on a DL380G3 and it never froze. > > And at the end let me say that personally I like the DL380s. You can > even resize Logical Drives of the RAID (e.g. when putting in > some new harddisc) without shutting down the server! > > Good Luck! > > Christoph > Thanks for all of the replies, both on and off list. I should have also mentioned that I'll be installing Debian Stable on it. Are you using Debian or something else? There seems to be an issue with the G4 and the Digium TE cards, so I'm unsure about placing the order at this time. I've been eyeing the Sangoma A104 card and the ipVolution TDM120 (not out yet, I believe) as an alternative. I have a service contract with an HP provider, so I'd prefer to stick with an HP server, if at all possible. However, if the T1 interfaces cause issues, there may be no choice. -- Dana ___ 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] HP ProLiant server for Asterisk
Thanks, I've been reading that. I'm pretty sure that HP is our only option here, so maybe I'll try getting quotes on the other models. -- Dana On Fri, 4 Feb 2005 17:31:26 +0200, Edge Bisset <[EMAIL PROTECTED]> wrote: > Hi Dana > > The DL380 G4 may not be the best choice; there has been a lot of talk on > the forum about problems with the DL380 G4 & the TE410P. > > See this thread: > http://lists.digium.com/pipermail/asterisk-users/2005-January/081544.htm > l > > Cheers, > Edge. > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson > Sent: Friday, February 04, 2005 5:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] HP ProLiant server for Asterisk > > I'm looking at ordering a server from HP. I checked around on Google and > found in the Wiki that the ProLiant DL380 is supposed to be known to > work with *. > > I'm going to get a price quote on the following setup: > > HP ProLiant DL380 G4 Server w/ the following options: > Intel Xeon 3.20GHz/1MB > 2GB REG PC2-3200 (2 X 1GB) > HP ProLiant Battery Backed Write Cache Enabler for SA6i > RAID 1 drive set > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive > Hot Plug Redundant Power Supply Module HP Redundant Fan Option Kit (3 > fans) 1.44MB Floppy Disk Drive Slimline 24X CD-ROM > > Can anyone comment any further on this system? Do you think I would make > a wise choice to order this? I will be putting a 4-port Digium TE card > in it. > > -- > Dana ___ 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] HP ProLiant server for Asterisk
I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. I'm going to get a price quote on the following setup: HP ProLiant DL380 G4 Server w/ the following options: Intel Xeon 3.20GHz/1MB 2GB REG PC2-3200 (2 X 1GB) HP ProLiant Battery Backed Write Cache Enabler for SA6i RAID 1 drive set HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive Hot Plug Redundant Power Supply Module HP Redundant Fan Option Kit (3 fans) 1.44MB Floppy Disk Drive Slimline 24X CD-ROM Can anyone comment any further on this system? Do you think I would make a wise choice to order this? I will be putting a 4-port Digium TE card in it. -- Dana ___ 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] Forwarding voicemail messages
On Thu, 3 Feb 2005 10:44:18 -0800, Gareth J. Greenaway <[EMAIL PROTECTED]> wrote: > I am encountering a problem with the voicemail portion of asterisk, when > someone goes to forward a voicemail and they choose the option to prepend the > message with a greeting, the call never ends, eventually this causes the > asterisk process to crash as the message file reaches 2GB. Is this a known > bug or perhaps something not setup correctly on my end. Thanks. > > Gareth J. Greenaway Does this happen if you specify a maximum length for messages (maxmessage = 180, for example, in voicemail.conf)? -- Dana ___ 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] Developing an IP Phone
I would be interested in this. I have Delphi 6 I believe. I'm a little rusty, but I'd like to make a really basic client. If you have any pointers along with that DLL, that would be fantastic. -- Dana On Tue, 1 Feb 2005 08:28:39 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > That's fine if you want to develop an SIP phone, but if you want an > IAX one, you can take iaxclient and compile it as a DLL. > > I did that and now I'm using it with Delphi. My phone is almost done :) > > I'll post it here when it's ready (really soon) > > N.B.: if somebody want the DLL, I'll be glad to share ___ 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
