Re: [Asterisk-Users] CID Matches On Incoming BroadVoice??? {Scanned}

[David Shaw]

Where did you get the patch? 

Thanks, David





On Tue, 2005-07-19 at 17:09 -0400, BJ Weschke wrote:
>  Tried that. It doesn't work. * ends up matching the peer/extension to
> the last registration that went through. That's why I had to patch
> chan_sip.c
> 
> 
> On 7/19/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > Wouldn't it have been easier to do...
> > 
> > register => user:[EMAIL PROTECTED]/1234
> > register => user:[EMAIL PROTECTED]/2345
> > 
> > And then create a dialplan for extensions 1234, 2345, etc?
> > 
> > > -Original Message-
> > > From: BJ Weschke [mailto:[EMAIL PROTECTED]
> > > Sent: Tuesday, July 19, 2005 8:11 AM
> > > To: Nate Kapi; Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] CID Matches On Incoming BroadVoice???
> > >
> > >
> > >  The way I fixed this with another provider that had similar
> > > behavior was to patch chan_sip.c so that it pulled out the
> > > DNIS value from the
> > > to: tag in the SIP header and then threw that it into the
> > > DNID channel variable. Then, I took the common extension ('s'
> > > in your case) and did a Goto with the value now in DNID to
> > > get to the appropriate place in the dial plan based on the
> > > number dialed.
> > >
> > > On 7/19/05, Nate Kapi <[EMAIL PROTECTED]> wrote:
> > > > I have been trying to make Broadvoice match incoming Caller
> > > ID and do
> > > > specific things based on the number received, but due to Broadvoice
> > > > requiring the "s" to start off an incoming extension, I cannot get
> > > > this to work. Has anyone been able to do this? Here are
> > > some examples
> > > > of my setup:
> > > >
> > > > from sip.conf:
> > > > context=broadvoice-incoming
> > > >
> > > > from extensions.conf
> > > > [broadvoice-incoming]
> > > > exten => s/760899,1,AGI(db.agi)
> > > > exten => s/760899,2,Answer
> > > > exten => s/760899,3,Goto(menu,s,1)
> > > >
> > > > exten => s,1,Answer
> > > > exten => s,2,Goto(differentmenu,s,1)
> > > >
> > > > I want calls from one NPA-NXX- to go to one menu and all other
> > > > calls to another. It has never worked for me though.
> > > >
> > > > Any ideas? Thanks! ___
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
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> > > >
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/aster> isk-users
> > > To
> > > UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > 
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Re: [Asterisk-Users] Did the Broadvoice patch break asterisk ? {Scanned}

[David Shaw]

I have never have the / work for me.

register=1234:[EMAIL PROTECTED]/ext

It has never worked for me.

David




On Wed, 2005-07-06 at 10:27 +0200, Christian Peter wrote:
> I dislike the statement in the bug reports "you can easily add / to
> your register statement as a workaround" because it simply does not work
> when having provider who redirects with sip 302 responses (eg.
> nikotel). 
> 
> Also can one tell me the reason for  /sipgateid  when registering at
> sipgate? It's not the extension but it does work.
> 
> Greetings
> 
> Christian Peter
> 
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Re: [Asterisk-Users] Did the Broadvoice patch break asterisk ? {Scanned}

[David Shaw]

I have 5 Lines with BV and I can't route the incoming calls because of
this problem. So yes I think its broken.

David

On Mon, 2005-07-04 at 19:30 +0200, Christian Peter wrote:
> > 
> > Yes, I know all of that, the problem is asterisk is NOT trying to
> > match anything except the IP from Broadvoiice. So all calls from BV
> > will be cached to the phone number and context of the first BV call.
> > Asterisk will not look at the phone number or context of the other BV
> > number when they come in.
> > 
> > So... Since Asterisk IS GETTING the phone number and context for the
> > first one that rings in, but IS NOT getting the phone number and
> > context for subsequent BV calls and INSTEAD caches and uses the info
> > from the first call, ASTERISK IS BROKEN.
> > 
> > Think about it and I think anyone would agree this is not a good thing
> > (can't send calls to correct destination based on context, can't bill
> > the proper partry for their phone service).
> > 
> > Thanks
> > 
> 
> Hi,
> 
> is that the reason why asterisk doesn't work with multiple accounts at
> the same provider? I read somewhere about it and indeed have problems
> with calls from two nikotel accounts.
> 
> If that's the case I'll vote for broken, too.
> 
> Christian Peter
> 
> 
> 
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Re: [Asterisk-Users] Computer to use {Scanned}

[David Shaw]

I'm using a cheap PCChip Mboard and an AMD 2800 Simpron 1Gb of RAM in a
cheap 2U case. I have two X100P and 5 Broadvoice lines. I have about 20
IP phones and softphones.

Works great.

73's David
KE6UPI

On Fri, 2005-07-01 at 09:36 -0700, Robert Goodyear wrote:
> On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote:
> 
> > Robert Goodyear wrote:
> >
> >> I'm sure you really only want to know about the absence of problems.  
> >> From watching this list for 6 months it seems the SuperMicro products 
> >> are most lauded and have exhibited no hardware conflicts. Various 
> >> votes on Dell products, so you're probably best to stay away, even 
> >> though I've got five installs with TE110Ps in them that have never 
> >> missed a beat -- Dimension boxes, not PowerEdge.
> >
> > The SuperMicro Xeon board we tried failed miserably with both the 
> > T100P and TE110P.  It had the ServerWorks IDE Chipset, which I suspect 
> > was the problem.
> >
> > -- 
> > Eric Wieling * BTEL Consulting * 504-210-3699 x2120
> 
> Bummer! I thought I'd heard all good things about them... sorta like 
> VoIP providers; as soon as everyone agrees things are OK, something 
> goes awry!
> 
> -Rob.
> 
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[Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?

[David Shaw]

Hello I saw Ver1.2 is out. Whats new?

Thanks for the hard work, David

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[Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?

[David Shaw]

Hello I saw Ver1.2 is out. Whats new?

Thanks for the hard work, David

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[Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?

[David Shaw]

Hello I saw Ver1.2 is out. Whats new?

Thanks for the hard work, David

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RE: [Asterisk-Users] multiples broadvoice lines {Scanned}

[David Shaw]

Sorry Everyone, My mother past away this week. 

I see there might be some fixes for this. I will try them tonight. 

Thanks, David



On Thu, 2005-05-26 at 11:14 -0700, trixter http://www.0xdecafbad.com
wrote:
> On Thu, 2005-05-26 at 12:48 -0500, Jay Milk wrote:
> > Nothing wrong with putting them all in the same context and using Goto
> > -- in fact, I've been using that with nine SIP lines from three
> > different providers and a dozen incoming DIDs from two IAX providers.
> > Why, you ask?  Because you have your ALL call-distribution nicely
> > contained in a single file -- extensions.conf.  
> 
> I never said there was anything wrong with that if that is what you
> choose to do, however I did say that if you do not choose to put them
> all in the same context and have them all go to different contexts
> instead asterisk ignores your feeble request and does what it wants.
> And that in my book qualifies as a bug.  
> 
> If I set a unique context for each account, the mere fact they are all
> from the same sip proxy should not override that.  It does not if they
> are from different proxies so it makes no sense that it does when they
> are the same proxy.  I think it was either a lazy programmer or a bad
> sort algorithm (perhaps an if that doesnt have enough compares for
> unique connection information?)
> 
> Granted this is a rare occurance for testing purposes, if a test case
> was not created to test for this problem specifically it would not be
> uncovered until someone used asterisk to try to do exactly this.  
> 
> I just feel that people should have choice, simple little freedoms to do
> their extensions.conf however they want, and not be forced to put them
> all in the same context if they do not want to.  Maybe my feelings on
> freedom and choice are too far out there and the better solution is to
> do it one way because that way is best for one person.
> 
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[Asterisk-Users] multiples broadvoice lines

[David Shaw]

Hello All, I have 4 Broadvoice lines. If I call any of the lines it
shows that is coming from the first line.

exaple

[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]

If I call X3 it shows that someone called X1.

ANY HELP Please.

I'm using [EMAIL PROTECTED] Ver 1

Thanks, David

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Re: [Asterisk-Users] asterisk with vonage linksys adapter? {Scanned}

[David Shaw]

No, but I got a SPA-2000 from ebay for $40 bucks.

David

On Sun, 2005-05-22 at 20:57 -0500, Matthew Boehm wrote:
> Short Answer: No
> 
> For the long answer: google.com
> 
> -Matthew
> 
> 
> > From: hank smith <[EMAIL PROTECTED]>
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > Date: Sun, 22 May 2005 13:31:12 -0700
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > Subject: [Asterisk-Users] asterisk with vonage linksys adapter?
> > 
> > hello do you know if vonage unlocks there linksys adapter to use with other
> > providers? I want to use my ixisting vonage adapter with asterisk and cancil
> > my vonage service.
> > thanks
> > hank
> > 
> > email:
> > [EMAIL PROTECTED]
> > gmail:
> > [EMAIL PROTECTED]
> > msn messenger:
> > [EMAIL PROTECTED]
> > aim:
> > hanksmith5
> > skype:
> > hanksmith5
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
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RE: [Asterisk-Users] 2 Asterisk boxes sharing dial plans. {Scanned}

[David Shaw]

Well not what I was thinking. I would like to share the outbound trunks.
One server needs an extra line it could use the other server.

Thanks, David


On Sun, 2005-05-22 at 16:42 -0400, Race Vanderdecken wrote:
> Could you set up an NFS directory that is shared between the servers?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
> Sent: Sunday, May 22, 2005 9:50 AM
> To: Asterisk Users Mailing List
> Subject: [Asterisk-Users] 2 Asterisk boxes sharing dial plans.
> 
> Hello All, I have two asterisk boxes. 1 for home and 1 for work/ham
> radio club. I have 2 SIP trunks on each server. What is the best way to
> share the trunks?
> 
> Thanks, David
> 
> PS FWDOUT is great!!!
> 
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[Asterisk-Users] 2 Asterisk boxes sharing dial plans.

[David Shaw]

Hello All, I have two asterisk boxes. 1 for home and 1 for work/ham
radio club. I have 2 SIP trunks on each server. What is the best way to
share the trunks?

Thanks, David

PS FWDOUT is great!!!

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Re: [Asterisk-Users] Asterisk@Home 0.9 released {Scanned}

[David Shaw]

Thanks for your hard work.

Thanks, David


On Thu, 2005-04-14 at 07:15 -0700, [EMAIL PROTECTED] wrote:
> cool thanks for the update. next time please submit a
> bug to the [EMAIL PROTECTED] source forge project. Then it
> will get fixed in the next release. I had no idea this
> was broken.
> 
> 
> --- Time Bandit <[EMAIL PROTECTED]> wrote:
> > > More bug fixes. *69 works now. Cisco stuff works.
> > Lots
> > > of other fixes.
> > is phpconfig fixed ?
> > 
> > when editing a file, it doesn't show the list of
> > sections, it only list "Header"
> > 
> > What needs to be modified : 
> > In the function OC_readConfFile around line 131
> > change : 
> > $this->_OC_the_file[] = fgetc($file); to :
> > $this->_OC_the_file[] = fgets($file);
> > 
> > I have to manually edit it each time I install it.
> > 
> > Thanks for the great work so far
> > 
> > hth
> > ___
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> >
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> >
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> > 
> 
> 
>   
> __ 
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> Yahoo! Small Business - Try our new resources site!
> http://smallbusiness.yahoo.com/resources/
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Re: [Asterisk-Users] Asterisk Dual Servers {Scanned}

[David Shaw]

Hello, I did that using SIP. I setup an extension on both servers.
Register both servers using the new extensions. Then just route the
calls.

David


On Sat, 2005-04-09 at 15:01 -0300, Juan Luis Moyano wrote:
> Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and 
> what I want to get done is that if I dial 1X on SrvB the call must be 
> routed to extension X on SrvA and if I dial 2X on SrvA the call must be 
> routed to extension X on SrvB. I've read the www.voip-info.org wiki 
> abouta sterisk dual servers but couldn't succeed on get it working. 
> Perhaps someone that has a working dialplan similar to what I want to do 
> could post his config files or explain what to do. Thanks in advance.
> 
> -- 
> Juan Luis Moyano
> [EMAIL PROTECTED]
> 
> 
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Re: [Asterisk-Users] Asterisk@Home 0.8 released {Scanned}

[David Shaw]

I upgraded to .8 from .4 and I'm having problems with SPA-2000 ATA. Its
unable to re-register. I'm using the same hardware. I see the spa-2000
try using tcpdump. is there anything I should check???

Thanks, David



On Wed, 2005-03-30 at 07:50 -0800, [EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] 0.7 was a little buggy so we decided to
> release 0.8  It even has a few new features. 
> 
> AMP 1-10-007a
> SpanDSP 0.0.2pre11
> vsftpd server
> 
> If you have question about installing or configuring
> [EMAIL PROTECTED] please read the [EMAIL PROTECTED] Handbook.
> 
> http://asteriskathome.sourceforge.net/handbook/
> 
> If you cant find what you need try posting to our
> discussion forum.
> 
> http://sourceforge.net/forum/?group_id=123387
> 
> 
> 
>   
> __ 
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[Asterisk-Users] Asterisk@Home .8 SPA-2000

[David Shaw]

Hello All, I upgraded (installed) [EMAIL PROTECTED] .8 from .4. Now my
SPA-2000 will not stay registered. When the it needs to reregister it
may or may not. Line1 might be able too when Line2 can't and so on. When
on a call it will drop out. I did upgrade the SPA-2000. 

Any Help would be great..

Ping times are in the 0.733 ms

Thanks, David

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Re: [Asterisk-Users] Multiple BroadVoice Accounts Problem with Incoming calls {Scanned}

[David Shaw]

Tried that.. I added an extenstion and that didn't work. I think its on
Broadvoice's end but they say no..

David

On Wed, 2005-04-06 at 11:06 -0400, Randy Johnson wrote:
> I do not have my config files as this computer but it seems to me you 
> are missing something on the end
> 
> [EMAIL PROTECTED]::[EMAIL PROTECTED]/SOMETHING GOES HERE
> 
> [EMAIL PROTECTED]::[EMAIL PROTECTED]/SOMETHING GOES HERE1
> [EMAIL PROTECTED]::[EMAIL PROTECTED]/SOMETHING GOES HERE 2
> [EMAIL PROTECTED]::[EMAIL PROTECTED]/SOMETHING GOES HERE 3
> 
> 
> 
> 
> David Shaw wrote:
> 
> >Hello All, I have googled this problem and called Broadvoice and I still
> >haven't found an answer.
> >
> >I have 4 BroadVoice accounts on one Asterisk box. If you call lines 1-3
> >it rings on line 4. It makes it hard for routing inbound calls.. If I
> >change the order of the registered accounts in the sip.conf, then what
> >is last is used.
> >
> >Please Help..
> >
> >Thanks, David
> >
> >sip.conf
> >
> >[EMAIL PROTECTED]::[EMAIL PROTECTED]
> >[EMAIL PROTECTED]::[EMAIL PROTECTED]
> >[EMAIL PROTECTED]::[EMAIL PROTECTED]
> >[EMAIL PROTECTED]::[EMAIL PROTECTED]
> >
> >[BV-IN-1]
> >username=XX2076
> >user=XX2076
> >type=user
> >secret=
> >nat=yes
> >insecure=very
> >host=sip.broadvoice.com
> >fromdomain=sip.broadvoice.com
> >dtmfmode=inband
> >dtmf=inband
> >context=from-pstn
> >
> >Same for the next three.
> >Change account info.
> >
> >[BV-OUT-1]
> >username=XX2076
> >user=phone
> >type=peer
> >secret=
> >nat=yes
> >insecure=very
> >host=sip.broadvoice.com
> >fromuser=XX2076
> >fromdomain=sip.broadvoice.com
> >dtmfmode=inband
> >dtmf=inband
> >context=from-home
> >canreinvite=no
> >authname=XX2076
> >
> >Same for the next three.
> >Change account info.
> >
> >___
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> >  
> >
> 
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[Asterisk-Users] Multiple BroadVoice Accounts Problem with Incoming calls

[David Shaw]

Hello All, I have googled this problem and called Broadvoice and I still
haven't found an answer.

I have 4 BroadVoice accounts on one Asterisk box. If you call lines 1-3
it rings on line 4. It makes it hard for routing inbound calls.. If I
change the order of the registered accounts in the sip.conf, then what
is last is used.

Please Help..

Thanks, David

sip.conf

[EMAIL PROTECTED]::[EMAIL PROTECTED]
[EMAIL PROTECTED]::[EMAIL PROTECTED]
[EMAIL PROTECTED]::[EMAIL PROTECTED]
[EMAIL PROTECTED]::[EMAIL PROTECTED]

[BV-IN-1]
username=XX2076
user=XX2076
type=user
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn

Same for the next three.
Change account info.

[BV-OUT-1]
username=XX2076
user=phone
type=peer
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=XX2076
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-home
canreinvite=no
authname=XX2076

Same for the next three.
Change account info.

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[Asterisk-Users] *@Home .6 adding a outside number to a group

[David Shaw]

Hello, I tried to add an outside number (my cell phone) to the group. I
would like to have both extensions ring as well as my cell. I'm running
[EMAIL PROTECTED] V.6.

Thanks, David

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Re: [Asterisk-Users] VoIP service through Asterisk? {Scanned}

[David Shaw]

Hello, I'm a newbe as well. I used [EMAIL PROTECTED], 2 Broadvoice accounts
and one SPA-2000. Works great.

David


On Sun, 2005-03-20 at 13:52 -0600, Rich Adamson wrote:
> Top posting for consistency
> 
> I don't know what teliax has for international services/rates. I didn't
> have a need for those and didn't ask. Send them an email and ask.
> 
> It's fairly common knowledge that several of the itsp's are trying
> to profit from consolidating long distance, and some will offer rates
> and services that don't necessarily appear on their web pages. I'd
> suggest dropping some an email and see how quick they respond. Those
> that don't respond likely have other customer service problems as well.
> 
> 
> > I took a look at teliax.  The pay as you go plan appears not to include 
> > international dialing and the commercial plan is fixed price of $44.99 
> > per month capped at 500 international minutes a month.  Are you aware 
> > if they have international rates based on usage?
> > 
> > MARK.
> > 
> > Rich Adamson wrote:
> > 
> > Other than Broadvoice, are there any VoIP providers (Vonage, 
> > Packet8, 
> > etc) that can be hooked into Asterisk directly? I read about a 
> > scheme 
> > for Packet8 that involved routing it in through an analog 
> > connection 
> > on a FXO port...I'd rather have something I can connect in 
> > directly.
> >   
> > 
> > Save yourself some hassle, and use a provider which supports IAX. 
> > So far 
> > there is:
> > 
> > - iax.cc (haven't tried them)
> > - connect.voicepusle.com (haven't tried them)
> > - nufone.net (they're meant to be quite reliable - i use them)
> > - voipjet.com (their rates are very good - i use them too)
> > 
> > BTW, any of you guys know other providers like VoIPJet (i.e. low 
> > rates & 
> > low volume OK) ?
> > 
> > 
> > Add to that...
> >  teliax.com
> >  livevoip.com
> > both with excellent and timely customer support, "real" iax support, 
> > quality audio, and no jumping through BS for sip registration or
> > codec selection, good rates, etc.
> > 
> > ___
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> >   
> ---End of Original Message-
> 
> 
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Re: [Asterisk-Users] Broadvoice Multiple "lines" {Scanned}

[David Shaw]

I also have multiple line with Broadvoice. I would like to have each
incoming line ring a different extension and configure an internal user
to use his or her own broadvoice line..

Here is my sip.conf

register =>
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]


register =>
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]

[broadvoice1]
type=peer
username=XX
fromuser=XX
authuser=XX
secret=password
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-1]
type=friend
host=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
canreinvite=yes
nat=no
allow=ulaw

[broadvoice2]
type=peer
username=AA
fromuser=AA
authuser=AA
secret=password
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice2
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-2]
type=friend
host=sip.broadvoice.com
context=from-broadvoice2
dtmfmode=inband
canreinvite=yes
nat=no
allow=ulaw

so on so on

But all incoming calls on Broadvoice uses extensions.conf [broadvoice4]
or what evers the last line for broadvoice.


On Wed, 2005-03-09 at 18:53 -0600, James Taylor wrote:
> I configured this once now I forgot what I did.
> 
> Two Broadvoice accounts.
> Incoming is simple - just use the phone numbers.
> 
> Outgoing:
> 
> Dial out on a specific line
> and/or
> set up the groups and select the other "line" if the first one is busy?
> 
> 
> -- 
> James Taylor
> MetroTel
> 3505 Summerihll Road
> Suite 11
> Texarkana, Texas  75503
> 903-793-1956
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working now Re: [Asterisk-Users] Asterisk@Home .6 Problems with outbound calls using Broadvoice {Scanned}

[David Shaw]

On Thu, 2005-03-03 at 11:00 -0800, David Shaw wrote:
> Hello All, I have one X100P card for inbound calls. I use two Broadvoice
> SIP accounts for all my outbound calls. I'm unable to place calls using
> BV. Inbound BV calls are ok.
> 
> Verbosity is at least 3
> -- Executing Macro("SIP/201-365c", "dialout-default|XXX") in new
> stack
> -- Executing GotoIf("SIP/201-365c", "1?4") in new stack
> -- Goto (macro-dialout-default,s,4)
> -- Executing GotoIf("SIP/201-365c", ""The Shaws"?6") in new stack
> -- Executing SetCallerID("SIP/201-365c", ""The Shaws" ")
> in new stack
> -- Executing Dial("SIP/201-365c", "SIP/BV-OUT-2/XXX") in new
> stack
> -- Called BV-OUT-2/XXX
> -- Got SIP response 604 "Does not exist anywhere" back from
> 147.135.0.128
>   == No one is available to answer at this time
> -- Executing Congestion("SIP/201-365c", "") in new stack
>   == Spawn extension (macro-dialout-default, s, 7) exited non-zero on
> 'SIP/201-365c' in macro 'dialout-default'
>   == Spawn extension (from-internal, XXX, 1) exited non-zero on
> 'SIP/201-365c'
> -- Executing Macro("SIP/201-365c", "hangupcall") in new stack
> -- Executing ResetCDR("SIP/201-365c", "w") in new stack
> -- Executing NoCDR("SIP/201-365c", "") in new stack
> -- Executing Wait("SIP/201-365c", "5") in new stack
>   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
> 'SIP/201-365c' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/201-365c'
> 
> Thanks, David
> 
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[Asterisk-Users] Asterisk@Home .6 Problems with outbound calls using Broadvoice

[David Shaw]

Hello All, I have one X100P card for inbound calls. I use two Broadvoice
SIP accounts for all my outbound calls. I'm unable to place calls using
BV. Inbound BV calls are ok.

Verbosity is at least 3
-- Executing Macro("SIP/201-365c", "dialout-default|XXX") in new
stack
-- Executing GotoIf("SIP/201-365c", "1?4") in new stack
-- Goto (macro-dialout-default,s,4)
-- Executing GotoIf("SIP/201-365c", ""The Shaws"?6") in new stack
-- Executing SetCallerID("SIP/201-365c", ""The Shaws" ")
in new stack
-- Executing Dial("SIP/201-365c", "SIP/BV-OUT-2/XXX") in new
stack
-- Called BV-OUT-2/XXX
-- Got SIP response 604 "Does not exist anywhere" back from
147.135.0.128
  == No one is available to answer at this time
-- Executing Congestion("SIP/201-365c", "") in new stack
  == Spawn extension (macro-dialout-default, s, 7) exited non-zero on
'SIP/201-365c' in macro 'dialout-default'
  == Spawn extension (from-internal, XXX, 1) exited non-zero on
'SIP/201-365c'
-- Executing Macro("SIP/201-365c", "hangupcall") in new stack
-- Executing ResetCDR("SIP/201-365c", "w") in new stack
-- Executing NoCDR("SIP/201-365c", "") in new stack
-- Executing Wait("SIP/201-365c", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/201-365c' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/201-365c'

Thanks, David

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Re: [Asterisk-Users] Vonage, broadvoice et al {Scanned}

[David Shaw]

I have two home accounts with Vonage and I allow all the family to use
Vonage with there extensions.

David


On Fri, 2005-02-18 at 08:39 -0500, Pedro wrote:
> Vonage, to my knowledge, does not let you connect your own SIP device
> to their service.  They provide their own IAD.
> 
> As for Broadvoice, I know people that have successfully deployed
> asterisk with many people sharing the same account.
> 
> - Pedro
> 
> 
> On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn <[EMAIL PROTECTED]> wrote:
> > Hi all,
> > 
> > I'm just wondering about these VoIP services -- do you have to sign up one
> > account -per- client that will be using the service? I've got multiple
> > extensions behind my Asterisk box, and I want to be able to allow all my 
> > staff
> > to place calls via the provider.
> > 
> > So if I sign up for one account, will multiple users behind my Asterisk box 
> > be
> > able to make calls, using that same account, at the same time? Or do these
> > providers typically only allow one call to be in place at any point in time?
> > 
> > Thanks in advance.
> > 
> > Flynn
> > 
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-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] HELP!!!!!!!! {Scanned}

[David Shaw]

If your X-ten phones are on the same lan as asterisk then try nat=no.

David

On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote:
> My sip.conf file;
> 
> [luke]
> type=friend
> host=dynamic
> username=luke
> secret=luke
> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
> dtmfmode=rfc2833
> mailbox=202 ; Mailbox for message waiting indicator
> allow=all
> context=sip
> callerid="luke" <2123>
> nat=yes
> 
> 
> [mike]
> type=friend
> host=dynamic
> username=mike
> secret=badru
> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
> dtmfmode=rfc2833
> mailbox=203 ; Mailbox for message waiting indicator
> context=sip
> callerid="mike" <2125>
> nat=yes
> 
> 
> [juki]
> type=friend
> host=dynamic
> username=juki
> secret=juki
> dtmfmode=rfc2833
> mailbox=204 ; Mailbox for message waiting indicator
> allow=all
> context=sip
> callerid="juki" <2125>
> nat=yes
> 
> 
> plus my extensions.conf file;
> 
> exten => 202,1,Dial(SIP/luke,20,tr)
> exten => 202,2,VoiceMail,u202
> exten => 202,102,VoiceMail,b202
> exten => 203,1,Dial(SIP/mike,20,tr)
> exten => 203,2,VoiceMail,u203
> exten => 203,102,VoiceMail,b203
> exten => 204,1,Dial(SIP/juki,20,tr)
> exten => 204,2,VoiceMail,u204
> exten => 204,102,VoiceMail,b204
> 
> Hope this provides a little bit more info.
> 
> 
> > I new to this as will. But add more info like your sip.conf file.
> >
> > David
> >
> >
> > On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote:
> >> Hi,
> >>
> >> I have installed two X-Lite phones and theyÃââre able to login
> >> successfully. The two phones plus the Asterisk system are all on the
> >> same LAN with private addresses assigned to each of them.  When a call
> >> is initiated and is picked up on the other end, there is completely no
> >> sound at all (as in the line goes dead). The codecs set in the
> >> softphones are g711u, g711a, GSM, iLBC and SPX.
> >>
> >> From the Asterisk CLI I see the following errors;
> >>
> >> i)Unknown RTP codec 72 received
> >>
> >> ii)  RFC3389 support incomplete
> >>
> >> Anyone got ideas on how I can go about this?
> >>
> >> Thanks in advance.
> >>
> >> Julius Kidubuka
> >>
> >> "When you do the common things in life in an uncommon way, you will
> >> command the attention of the world"
> >>
> >>
> >>
> >>
> >>
> >> --
> >> This message has been scanned for viruses and
> >> dangerous content by MailScanner, and is
> >> believed to be clean.
> >> MailScanner thanks transtec Computers for their support.
> >> Plase contact [EMAIL PROTECTED] if you have questions about
> >> this email.
> >> ___
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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> > --
> > David Shaw <[EMAIL PROTECTED]>
> >
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> 
> 
> -- 
> Rgds,
> Julius Kidubuka.
> "My advice to you is get married: if you find a good wife you'll be happy;
> if not, you'll become a philosopher."
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> Spam detection software, running on the system "zeus.avanzada7.com", has
> identified this incoming email as possible spam.  The original message
> has been attached to this so you can view it (if it isn't spam) or label
> similar future email.  If you have any questions, see
> the administrator of that system for details.
> 
> Content preview:  My sip.conf file; [luke] type=friend host=dynamic 
>   username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband, 
>   rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message 
>   waiting indicator allow=all context=sip callerid="luke" <2123> nat=yes 
>   [...] 
> 
> Content analysis details:   (0.5 points, 5.0 required)
> 
>  pts rule name  description
>  -- --
>  0.1 FORGED_RCVD_HELO   Received: contains a forged HELO
>  0.4 PLING_PLINGSubject has lots of exclamation marks
> 
> 
-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk@Home 0.6 Released {Scanned}

[David Shaw]

Can you add ez-ipupdate to the web interface and maybe PPPOE
configuration

http://ez-ipupdate.com/

Thanks, David

You ask why?? I run this from home..

On Wed, 2005-02-16 at 10:59 -0800, [EMAIL PROTECTED] wrote:
> New features include Festival text to speech and a new
> Web Conferencing GUI. There are also numerous small
> fixes and enhancements.
> 
> http://asteriskathome.sourceforge.net/
> 
> 
> 
>   
> __ 
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> All your favorites on one personal page  Try My Yahoo!
> http://my.yahoo.com 
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-- 
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Re: [Asterisk-Users] Monitor does not like variable subsitutions {Scanned}

[David Shaw]

Here is what I use for outbound calls.

exten => _1NXXNXX,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => _1NXXNXX,2,Monitor(wav,${CALLFILENAME},m)
exten => _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})

David


On Wed, 2005-02-16 at 07:57 -0800, Jason Goecke wrote:
> Hello,
> 
> I have been attempting to get the Monitor function to
> accept a loal variable substitution in order to use
> the same filename later in the same context.  Monitor
> does not appear to like it, as it attempts to use
> wav|filename as the recording type, as opposed to just
> wav.
> 
> Here is what I get if I just supply a filename
> directly (it works fine):
> 
> --context-
> exten => _9X.,3,Monitor(wav|recording|m)
> --context-
> 
> --CLI-
> -- Executing SetVar("SIP/3004-275c",
> "REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10")
> in new stack
> -- Executing Monitor("SIP/3004-275c",
> "wav|recording|m") in new stack
> -- Executing AGI("SIP/3004-275c", "outbound.agi")
> in new stack
> --CLI-
> 
> Here is what I get when I attempt to to variable
> substituion for the filename:
> 
> 
> --context-
> exten =>
> _9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
> exten => _9X.,3,Monitor(wav|${FILENAME}|m)
> --context-
> 
> --CLI-
> -- Executing SetVar("SIP/3004-da21",
> "REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35")
> in new stack
> -- Executing Monitor("SIP/3004-da21",
> "wav|rec_to_448704386865_at_16022005-16:56:35|m") in
> new stack
> Feb 16 16:56:35 WARNING[17028]: file.c:934
> ast_writefile: No such format
> 'wav|rec_to_448704386865_at_16022005-16'
> Feb 16 16:56:35 WARNING[17028]: res_monitor.c:154
> ast_monitor_start: Could not create file
> /var/spool/asterisk/monitor/m-in
> Feb 16 16:56:35 WARNING[17028]: res_monitor.c:300
> ast_monitor_change_fname: Cannot change monitor
> filename of channel SIP/3004-da21 to m, monitoring not
> started-- Executing AGI("SIP/3004-da21",
> "outbound.agi") in new stack
> --CLI-
> 
> I do believe that I had this working before (I am
> running the CVS HEAD from yesterday).
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Re: [Asterisk-Users] HELP!!!!!!!! {Scanned}

[David Shaw]

I new to this as will. But add more info like your sip.conf file.

David


On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote:
> Hi,
> 
> I have installed two X-Lite phones and theyâre able to login
> successfully. The two phones plus the Asterisk system are all on the
> same LAN with private addresses assigned to each of them.  When a call
> is initiated and is picked up on the other end, there is completely no
> sound at all (as in the line goes dead). The codecs set in the
> softphones are g711u, g711a, GSM, iLBC and SPX.
> 
> From the Asterisk CLI I see the following errors;
> 
> i)Unknown RTP codec 72 received
> 
> ii)  RFC3389 support incomplete
> 
> Anyone got ideas on how I can go about this?
> 
> Thanks in advance.
> 
> Julius Kidubuka
> 
> "When you do the common things in life in an uncommon way, you will
> command the attention of the world"
> 
>  
> 
> 
> 
> -- 
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> dangerous content by MailScanner, and is 
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Re: [Asterisk-Users] Help With Broadvoice {Scanned}

[David Shaw]

I have two BV accounts. For host= I used sip.broadvoice.com at first.
Then I changed it to the faster proxy. My sip.conf looks just like this
without all the extensions.

sip.conf
[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0  ; Address to bind SIP channel to
;context=default   ; Default context for incoming calls

register => phone-number1:[EMAIL PROTECTED]
register => phone-number2:[EMAIL PROTECTED]

[broadvoice1]
type=friend
username=phone-number
fromuser=phone-number
secret=sip-passwd
host=sip.broadvoice.com ;you need to check BV proxy
fromdomain=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-1]
type=friend
host=sip.broadvoice.com ;you need to check BV proxy
context=from-broadvoice1
dtmfmode=inband
canreinvite=no
nat=no
allow=ulaw

[broadvoice2]
type=friend
username=phone-number2
fromuser=phone-number2
secret=sip-passwd2
host=sip.broadvoice.com ;you need to check BV proxy
fromdomain=sip.broadvoice.com
context=from-broadvoice2
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-2]
type=friend
host=sip.broadvoice.com ;you need to check BV proxy
context=from-broadvoice2
dtmfmode=inband
canreinvite=no
nat=no
allow=ulaw

:extensions

[100]
type=friend
username=100
secret=passwd
host=dynamic
mailbox=100
callerid="Kelli" <100>
nat=no

[101]
type=friend
username=101
secret=passwd
host=dynamic
callerid="David" <101>
mailbox=101
nat=no


On Wed, 2005-02-16 at 10:19 -0800, Max Clark wrote:
> David,
> 
> Thanks for the reply. Just to clarify, is the register and first 
> type=friend block all within the [general] section of sip.conf?
> 
> Thanks,
> Max
> 
>Max Clark
>    max [at] clarksys.com
>http://www.clarksys.com
> 
> 
> David Shaw wrote:
> > Here is my conf files.
> > 
> > sip.conf
> > 
> > register => phone#:sip/[EMAIL PROTECTED]
> > 
> > type=friend
> > username=phone#
> > fromuser=phone#
> > secret=sip/passwd
> > host=sip.broadvoice.com
> > fromdomain=sip.broadvoice.com
> > context=from-broadvoice1
> > dtmfmode=inband
> > disallow=all
> > allow=ulaw
> > canreinvite=no
> > nat=no
> > 
> > [bv-in-1]
> > type=friend
> > host=sip.broadvoice.com
> > context=from-broadvoice1
> > dtmfmode=inband
> > canreinvite=no
> > nat=no
> > allow=ulaw
> > 
> > extensions.conf
> > exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
> > exten => _011,1,Dial(SIP/[EMAIL PROTECTED])
> > 
> > 
> > 
> > On Wed, 2005-02-16 at 00:52 -0500, Randy Johnson wrote:
> > 
> >>Remember that the password is not your broadvoice website password but 
> >>the one you need to get from broadvoice support.
> >>
> >>Randy
> >>
> >>
> >>Greg Hill wrote:
> >>
> >>
> >>>On Tue, 15 Feb 2005, Max Clark wrote:
> >>>
> >>> 
> >>>
> >>>
> >>>>I have experimented with several configs based on different pages and
> >>>>threads but nothing is working. How do I properly configure my
> >>>>broadvoice account?
> >>>>
> >>>>[general]
> >>>>register => [EMAIL PROTECTED]::[EMAIL PROTECTED]
> >>>>   
> >>>>
> >>>
> >>>the register I'm using looks like this:
> >>>register => 310584:@sip.broadvoice.com
> >>>
> >>> 
> >>>
> >>>
> >>>>[broadvoice]
> >>>>type=peer
> >>>>host=sip.broadvoice.com
> >>>>secret=
> >>>>fromuser=310584
> >>>>fromdomain=sip.broadvoice.com
> >>>>context=incoming
> >>>>dtmfmode=inband
> >>>>canreinvite=no
> >>>>nat=yes
> >>>>qualify=yes
> >>>>   
> >>>>
> >>>
> >>>try:
> >>>
> >>>[broadvoice]
> >>>type=peer
> >>>username=310584
> >>>secret=
> >>>host=sip.broadvoice.com
> >>>port=5060
> >>>context=incoming
> >>>fromuser=310584
> >>>fromdomain=sip.broadvoice.com
> >>>dtmfmode=inband
> >>>canreinvite=no
> >>>insecure=very
> >>>permit=147.135.8.128/32
> >>>qualify=yes
> >>>
> >>>and adjust your permit= line to match the IP of the BV proxy you've set in
> >>>your /etc/hosts

Re: [Asterisk-Users] Help With Broadvoice {Scanned}

[David Shaw]

Here is my conf files.

sip.conf

register => phone#:sip/[EMAIL PROTECTED]

type=friend
username=phone#
fromuser=phone#
secret=sip/passwd
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-1]
type=friend
host=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
canreinvite=no
nat=no
allow=ulaw

extensions.conf
exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _011,1,Dial(SIP/[EMAIL PROTECTED])



On Wed, 2005-02-16 at 00:52 -0500, Randy Johnson wrote:
> Remember that the password is not your broadvoice website password but 
> the one you need to get from broadvoice support.
> 
> Randy
> 
> 
> Greg Hill wrote:
> 
> >On Tue, 15 Feb 2005, Max Clark wrote:
> >
> >  
> >
> >>I have experimented with several configs based on different pages and
> >>threads but nothing is working. How do I properly configure my
> >>broadvoice account?
> >>
> >>[general]
> >>register => [EMAIL PROTECTED]::[EMAIL PROTECTED]
> >>
> >>
> >
> >the register I'm using looks like this:
> >register => 310584:@sip.broadvoice.com
> >
> >  
> >
> >>[broadvoice]
> >>type=peer
> >>host=sip.broadvoice.com
> >>secret=
> >>fromuser=310584
> >>fromdomain=sip.broadvoice.com
> >>context=incoming
> >>dtmfmode=inband
> >>canreinvite=no
> >>nat=yes
> >>qualify=yes
> >>
> >>
> >
> >try:
> >
> >[broadvoice]
> >type=peer
> >username=310584
> >secret=
> >host=sip.broadvoice.com
> >port=5060
> >context=incoming
> >fromuser=310584
> >fromdomain=sip.broadvoice.com
> >dtmfmode=inband
> >canreinvite=no
> >insecure=very
> >permit=147.135.8.128/32
> >qualify=yes
> >
> >and adjust your permit= line to match the IP of the BV proxy you've set in
> >your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com).
> >
> >Try a blend of this stuff with whatever the most recent recommendation on
> >their support page says.
> >
> >Greg
> >
> >
> >___
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> >
> >  
> >
> 
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[Asterisk-Users] Making ZAP Trunk groups

[David Shaw]

Hello, I installed [EMAIL PROTECTED] .5 and I'm using AMP to configure it. I
have 4 X100P and two Broadvoice SIP accounts. Not all the X100P and BV
are used to outbound calls. How can I change the config files or what
should I change so just the trunks and bv lines I want to use are used.
-- 
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Re: [Asterisk-Users] Asterisk@Home .5 Setup help with 4 X100P {Scanned} {Scanned}

[David Shaw]

Cool, This is the output

Chan Extension  Context Language   MusicOnHold
 pseudoline4  default
  1line1  default
  2line2  default
  3line3  default
  4line4  default
Verbosity is at least 3

Quick test dialing in showed that worked... 

Thanks very much I'm still new., David

On Tue, 2005-02-15 at 14:01 -0800, [EMAIL PROTECTED] wrote:
> Have you tried the new auto-config script? Type
> help-aah for more info.
> 
> --- goldhorse <[EMAIL PROTECTED]> wrote:
> 
> > If you have 4 fxo cards I think you should modify
> > zapata.conf to
> > channel => 1-4
> > I haven't used [EMAIL PROTECTED] so I don't know how it
> > starts asterisk, 
> > but just a thought,
> > does it do modprobe wcfxo then ztcfg? Try doing this
> > at the terminal, 
> > if something's
> > wrong you should get error messages.
> > 
> > Hope this helps.
> > 
> > Daniel
> > 
> > 
> > On 2005/02/16, at 4:18, Rich Adamson wrote:
> > 
> > >
> > >> I removed the ; before channel => 1 and restarted
> > * and I get an 
> > >> error.
> > >>
> > >> Ouch ... error while writing audio data: : Broken
> > pipe
> > >>
> > >> ?
> > >>
> > >> I then replaced the ; and its starts??
> > >>
> > >> I'm upgrading from .4 to .5
> > >>
> > >> .4 was running
> > >>
> > >> Thanks, David
> > >>
> > >>
> > >>
> > >> On Tue, 2005-02-15 at 12:46 -0500, dean collins
> > wrote:
> > >>> David,
> > >>> You will need to go into the Zapata.conf file
> > and remove the ; before
> > >>> channel => 1
> > >>>
> > >>> (modify it to suit however you want to set up
> > your 4 X100p lines.
> > >>>
> > >>> The outgoing lines you see in FOP are only
> > example lines, you will 
> > >>> need
> > >>> to modify FOP to get rid of them etc.
> > >>>
> > >>>
> > >>> -Original Message-
> > >>> From: [EMAIL PROTECTED]
> > >>> [mailto:[EMAIL PROTECTED]
> > On Behalf Of David 
> > >>> Shaw
> > >>> Sent: Tuesday, February 15, 2005 11:02 AM
> > >>> To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > >>> Subject: [Asterisk-Users] [EMAIL PROTECTED] .5 Setup
> > help with 4 X100P
> > >>>
> > >>> Hello All, I installed [EMAIL PROTECTED] .5 last
> > night. I was able to
> > >>> configure some extensions for the house and they
> > work fine. I just 
> > >>> can't
> > >>> make inbound and/or outbound calls. The Flash
> > Operator Panel shows 
> > >>> four
> > >>> external icons and my new extensions.
> > >>>
> > >>> I have four X100P and two Broadvoice sip
> > accounts.
> > >>>
> > >
> > >
> > > The broken pipe message usually means something is
> > misconfigured. Check
> > > your zapata.conf against the following:
> > >
> > > context=inbound-bus
> > > signalling=fxs_ks
> > > echocancel=yes
> > > echotraining=800
> > > echocancelwhenbridged=no
> > > rxgain=3.0
> > > txgain=0.0
> > > immediate=yes
> > > channel => 1
> > >
> > > After making changes to zapata.conf, be sure to
> > stop and start *.
> > > Don't use reload on these type changes.
> > >
> > >
> > >
> > > ___
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> > >
> >
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> > >   
> >
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> > >
> > 
> > ___
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> >
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> >
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> > 
> 
> 
> 
>   
> __ 
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-- 
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RE: [Asterisk-Users] Asterisk@Home .5 Setup help with 4 X100P {Scanned}

[David Shaw]

I removed the ; before channel => 1 and restarted * and I get an error.

Ouch ... error while writing audio data: : Broken pipe

?

I then replaced the ; and its starts??

I'm upgrading from .4 to .5

.4 was running

Thanks, David



On Tue, 2005-02-15 at 12:46 -0500, dean collins wrote:
> David,
> You will need to go into the Zapata.conf file and remove the ; before
> channel => 1
> 
> (modify it to suit however you want to set up your 4 X100p lines.
> 
> The outgoing lines you see in FOP are only example lines, you will need
> to modify FOP to get rid of them etc.
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
> Sent: Tuesday, February 15, 2005 11:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] [EMAIL PROTECTED] .5 Setup help with 4 X100P
> 
> Hello All, I installed [EMAIL PROTECTED] .5 last night. I was able to
> configure some extensions for the house and they work fine. I just can't
> make inbound and/or outbound calls. The Flash Operator Panel shows four
> external icons and my new extensions. 
> 
> I have four X100P and two Broadvoice sip accounts.
> 
> 
> Thanks, David
> -- 
> David Shaw <[EMAIL PROTECTED]>
> 
> ___
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[Asterisk-Users] Asterisk@Home .5 Setup help with 4 X100P

[David Shaw]

Hello All, I installed [EMAIL PROTECTED] .5 last night. I was able to
configure some extensions for the house and they work fine. I just can't
make inbound and/or outbound calls. The Flash Operator Panel shows four
external icons and my new extensions. 

I have four X100P and two Broadvoice sip accounts.


Thanks, David
-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Re: asterisk@home scary log {Scanned}

[David Shaw]

Cat your maillog. Grep out the msg ID.

cat /var/log/maillog | grep j1A1U7Q1010071


j1A1U7Q1010071 is the [EMAIL PROTECTED]

j1A1U7mf010088 is email from root to???

Have you checked root's email??

Your might want to edit 
/etc/aliases and forward root: [EMAIL PROTECTED]

Also check sendmail deamon ports.
cat /etc/mail/sendmail.cf | grep DaemonPortOptions

This mains only 127.0.0.1 can relay.
O DaemonPortOptions=Port=smtp,Addr=127.0.0.1, Name=MTA

Good luck, David




On Thu, 2005-02-10 at 17:53 +0100, Bruno Hertz wrote:
> On Thu, 2005-02-10 at 11:09 -0500, Jason Stewart wrote:
> 
> > There's a chance that you may have been hacked, but the logs you post
> > look more like your mailserver is an open relay.
> 
> You sure? I run postfix myself and am not proficient in analyzing
> sendmail logs, but looking at those lines
> 
> Feb  9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088:
> from=<[EMAIL PROTECTED]>, size=329, class=0, nrcpts=1,
> msgid=<[EMAIL PROTECTED]>, proto=ESMTP,
> daemon=MTA, relay=asterisk1.local [127.0.0.1]
> Feb  9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071:
> [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00,
> xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1]
> [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for
> delivery)
> 
> 
> I find the relay (accepting host) is 127.0.0.1. So, even if ignoring
> the envelope 'from', there seems to be no doubt which host this mail was
> sent from.
> 
> Regards, Bruno.
> 
> 
> 
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Re: [Asterisk-Users] Broadvoice issues {Scanned}

[David Shaw]

I had problems as well. It was do to my sip.conf and extension.conf

Here are my conf files.

sip.conf


[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0  ; Address to bind SIP channel to
context=default   ; Default context for incoming calls


register => number:[EMAIL PROTECTED]/102
register => XX:[EMAIL PROTECTED]

[broadvoice]   ;--> This is what messed me up. This
type=friend   ; is up you will use in your exten
username=XX   ; line @broadvoice.
fromuser=XX
secret=sip-passwd
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes

[bv-in]
type=friend
host=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
allow=ulaw

extension.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for
demo
IAXINFO=guest   ; IAXtel
username/password
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)


[default]
include => DISA

exten => _0[1-9],1,Background,pls-hold-while-try
exten => _0[1-9],2,Dial(SIP/[EMAIL PROTECTED])
exten => _1XX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _911,1,Goto(911,911,1)
exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _NXX,2,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _8500,1,Goto(VoiceMail,8500,1)

[from-broadvoice]
exten => s,1,Wait(2)
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,5
exten => s,5,Background,welcome
exten => s,6,Background,if-u-know-ext-dial
exten => 8501,1,Goto(DISA,8501,1)
exten => 8510,1,Goto(8510,8510,1)
exten => i,1,Playback,pbx-invalid
exten => i,2,Goto,s|6
exten => t,1,Goto,0|1
exten => _##,1,Hangup

[from-pbx1]
exten => 8510,1,Dial(SIP/8510,10)
exten => 8510,2,Voicemail,u8510

[8510]
exten => 8510,1,Wait(2)
exten => 8510,2,Answer
exten => 8510,3,Background,pls-hold-while-try
exten => 8510,4,Dial(SIP/8510,10)
exten => 8510,5,Background,pls-hold-while-try
exten => 8510,6,Dial(SIP/[EMAIL PROTECTED],15)
exten => 8510,7,Background,tt-somethingwrong
exten => 8510,8,Voicemail,u8510

[VoiceMail]
exten => 8500,1,VoicemailMain

[DISA]
exten => 8501,1,Answer
exten => 8501,2,Wait,1
exten => 8501,3,DigitTimeout,5
exten => 8501,4,ResponseTimeout,10
exten => 8501,5,Authenticate(XXX)
exten => 8501,6,DISA,no-password|default
exten => i,7,Hangup

[911]
exten => _911,1,Background,no-911-1
exten => _911,2,Dial(SIP/8510,20)
exten => _911,3,Goto(default,911,1)


I hope this helps.

David


On Mon, 2005-02-07 at 14:36 -0800, Luki wrote:
> > You are probably using your website password
> The password used for registering is the same you use for outgoing
> calls -- yes, it's different from your "portal" password. So if you
> can register and receive calls, you have the password you need.
> 
> Double check that you use the section name from sip.conf in your dial
> plan, and that you have the correct password as well as the fromuser
> and username set in the broadvoice section in sip.conf.
> 
> As Rich said before, post your relevant sip.conf (register statement
> and BV section) and your dialplan entry.
> 
> --Luki
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Broadvoice problems with outbound calls {Scanned}

[David Shaw]

Thanks that worked. 

Changing @sip.broadvoice.com to @broadvoice worked.

Now I can drop Vonage... Well maybe

Thanks, David 


On Thu, 2005-02-03 at 14:23 -0500, Randy Johnson wrote:
> > > exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
> 
> 
> the @sip.broadvoice.com should be   @broadvoice
> 
> broadvoice being the title inside your sip.conf
> 
> [broadvoice]
> 
> 
> Randy
> 
> - Original Message - 
> From: "David Shaw" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Thursday, February 03, 2005 1:26 PM
> Subject: Re: [Asterisk-Users] Broadvoice problems with outbound calls 
> {Scanned}
> 
> 
> > Here is the error I get.
> >
> > Asterisk Ready.
> > *CLI> -- Executing Dial("SIP/8510-5c56",
> > "SIP/[EMAIL PROTECTED]") in new stack
> >-- Called [EMAIL PROTECTED]
> >-- Got SIP response 604 "Does not exist anywhere" back from
> > 147.135.0.128
> >  == No one is available to answer at this time (1:0/0/0)
> >
> > David
> >
> > On Thu, 2005-02-03 at 10:02 -0800, David Shaw wrote:
> >> Broadvoice says that I'm registered. I force my outbound calls to BV and 
> >> I
> >> get a busy signal.
> >>
> >> David
> >>
> >>
> >>
> >> - Original Message -
> >> From: "Jay Milk" <[EMAIL PROTECTED]>
> >> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >> 
> >> Sent: Thursday, February 03, 2005 9:19 AM
> >> Subject: RE: [Asterisk-Users] Broadvoice problems with outbound calls
> >> {Scanned}
> >>
> >>
> >> > Try
> >> >
> >> > register => number:[EMAIL PROTECTED]
> >> >
> >> > Instead of that odd line you have down there.
> >> >
> >> > > -Original Message-
> >> > > From: Randy Johnson [mailto:[EMAIL PROTECTED]
> >> > > Sent: Thursday, February 03, 2005 8:44 AM
> >> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> > > Subject: Re: [Asterisk-Users] Broadvoice problems with
> >> > > outbound calls {Scanned}
> >> > >
> >> > >
> >> > > David,
> >> > >
> >> > > Here is why register line, not sure if it would be the same
> >> > > effect as yours:
> >> > >
> >> > > [EMAIL PROTECTED]:password:[EMAIL PROTECTED]
> >> > .broadvoice.com/s
> >> > >
> >> > >
> >> > > - Original Message -
> >> > > From: "David Shaw" <[EMAIL PROTECTED]>
> >> > > To: 
> >> > > Sent: Wednesday, February 02, 2005 4:49 PM
> >> > > Subject: [Asterisk-Users] Broadvoice problems with outbound
> >> > > calls {Scanned}
> >> > >
> >> > >
> >> > > > Hello All, I sign up with $5.99 broadvoice plan. I made in and
> >> > > > outbound calls OK. I upgraded to unlimited world and now I have
> >> > > > problems with outbound calls. I called broadvoice and they
> >> > > said they
> >> > > > would get back it me.
> >> > > >
> >> > > > Here are my sip and extension files.
> >> > > >
> >> > > >
> >> > > > sip.conf
> >> > > >
> >> > > > register => XX:[EMAIL PROTECTED]
> >> > > >
> >> > > > [broadvoice]
> >> > > > type=friend
> >> > > > username=XX
> >> > > > fromuser=XX
> >> > > > secret=passwd
> >> > > > host=sip.broadvoice.com
> >> > > > fromdomain=sip.broadvoice.com
> >> > > > context=from-broadvoice
> >> > > > dtmfmode=inband
> >> > > > disallow=all
> >> > > > allow=ulaw
> >> > > > canreinvite=no
> >> > > > nat=yes
> >> > > >
> >> > > > [bv-in-1]
> >> > > > type=friend
> >> > > > host=sip.broadvoice.com
> >> > > > context=from-broadvoice
> >> > > > dtmfmode=inband
> >> > > > canreinvite=no
> >> > > > nat=yes
> >> > > > allow=ulaw
> >

Re: [Asterisk-Users] Broadvoice problems with outbound calls {Scanned}

[David Shaw]

Here is the error I get.

Asterisk Ready.
*CLI> -- Executing Dial("SIP/8510-5c56",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 604 "Does not exist anywhere" back from
147.135.0.128
  == No one is available to answer at this time (1:0/0/0)

David

On Thu, 2005-02-03 at 10:02 -0800, David Shaw wrote:
> Broadvoice says that I'm registered. I force my outbound calls to BV and I
> get a busy signal.
> 
> David
> 
> 
> 
> - Original Message -
> From: "Jay Milk" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Sent: Thursday, February 03, 2005 9:19 AM
> Subject: RE: [Asterisk-Users] Broadvoice problems with outbound calls
> {Scanned}
> 
> 
> > Try
> >
> > register => number:[EMAIL PROTECTED]
> >
> > Instead of that odd line you have down there.
> >
> > > -Original Message-
> > > From: Randy Johnson [mailto:[EMAIL PROTECTED]
> > > Sent: Thursday, February 03, 2005 8:44 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Broadvoice problems with
> > > outbound calls {Scanned}
> > >
> > >
> > > David,
> > >
> > > Here is why register line, not sure if it would be the same
> > > effect as yours:
> > >
> > > [EMAIL PROTECTED]:password:[EMAIL PROTECTED]
> > .broadvoice.com/s
> > >
> > >
> > > - Original Message -
> > > From: "David Shaw" <[EMAIL PROTECTED]>
> > > To: 
> > > Sent: Wednesday, February 02, 2005 4:49 PM
> > > Subject: [Asterisk-Users] Broadvoice problems with outbound
> > > calls {Scanned}
> > >
> > >
> > > > Hello All, I sign up with $5.99 broadvoice plan. I made in and
> > > > outbound calls OK. I upgraded to unlimited world and now I have
> > > > problems with outbound calls. I called broadvoice and they
> > > said they
> > > > would get back it me.
> > > >
> > > > Here are my sip and extension files.
> > > >
> > > >
> > > > sip.conf
> > > >
> > > > register => XX:[EMAIL PROTECTED]
> > > >
> > > > [broadvoice]
> > > > type=friend
> > > > username=XX
> > > > fromuser=XX
> > > > secret=passwd
> > > > host=sip.broadvoice.com
> > > > fromdomain=sip.broadvoice.com
> > > > context=from-broadvoice
> > > > dtmfmode=inband
> > > > disallow=all
> > > > allow=ulaw
> > > > canreinvite=no
> > > > nat=yes
> > > >
> > > > [bv-in-1]
> > > > type=friend
> > > > host=sip.broadvoice.com
> > > > context=from-broadvoice
> > > > dtmfmode=inband
> > > > canreinvite=no
> > > > nat=yes
> > > > allow=ulaw
> > > >
> > > > extensions.conf
> > > >
> > > > exten => _NXX,1,Dial(${TRUNKL2}/${EXTEN})
> > > > exten => _NXX,2,Dial(${TRUNKL3}/${EXTEN})
> > > > exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
> > > >
> > > > exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
> > > > exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
> > > > exten => _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})
> > > > exten => _01144XX,1,Dial(SIP/[EMAIL PROTECTED])
> > > > exten => _01144XX,2,Dial(${TRUNKL3}/${EXTEN})
> > > >
> > > > Thanks, David
> > > >
> > > > --
> > > > This message has been scanned for viruses and
> > > > dangerous content by KE6UPI, and is
> > > > believed to be clean.
> > > > KE6UPI thanks MailScanner for their support.
> > > > Please contact [EMAIL PROTECTED] if you have
> > > > questions about this email.
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ___
> > > Asterisk-Users mailing list
> > 

Re: [Asterisk-Users] Broadvoice problems with outbound calls {Scanned}

[David Shaw]

Broadvoice says that I'm registered. I force my outbound calls to BV and I
get a busy signal.

David



- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, February 03, 2005 9:19 AM
Subject: RE: [Asterisk-Users] Broadvoice problems with outbound calls
{Scanned}


> Try
>
> register => number:[EMAIL PROTECTED]
>
> Instead of that odd line you have down there.
>
> > -Original Message-
> > From: Randy Johnson [mailto:[EMAIL PROTECTED]
> > Sent: Thursday, February 03, 2005 8:44 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Broadvoice problems with
> > outbound calls {Scanned}
> >
> >
> > David,
> >
> > Here is why register line, not sure if it would be the same
> > effect as yours:
> >
> > [EMAIL PROTECTED]:password:[EMAIL PROTECTED]
> .broadvoice.com/s
> >
> >
> > - Original Message -
> > From: "David Shaw" <[EMAIL PROTECTED]>
> > To: 
> > Sent: Wednesday, February 02, 2005 4:49 PM
> > Subject: [Asterisk-Users] Broadvoice problems with outbound
> > calls {Scanned}
> >
> >
> > > Hello All, I sign up with $5.99 broadvoice plan. I made in and
> > > outbound calls OK. I upgraded to unlimited world and now I have
> > > problems with outbound calls. I called broadvoice and they
> > said they
> > > would get back it me.
> > >
> > > Here are my sip and extension files.
> > >
> > >
> > > sip.conf
> > >
> > > register => XX:[EMAIL PROTECTED]
> > >
> > > [broadvoice]
> > > type=friend
> > > username=XX
> > > fromuser=XX
> > > secret=passwd
> > > host=sip.broadvoice.com
> > > fromdomain=sip.broadvoice.com
> > > context=from-broadvoice
> > > dtmfmode=inband
> > > disallow=all
> > > allow=ulaw
> > > canreinvite=no
> > > nat=yes
> > >
> > > [bv-in-1]
> > > type=friend
> > > host=sip.broadvoice.com
> > > context=from-broadvoice
> > > dtmfmode=inband
> > > canreinvite=no
> > > nat=yes
> > > allow=ulaw
> > >
> > > extensions.conf
> > >
> > > exten => _NXX,1,Dial(${TRUNKL2}/${EXTEN})
> > > exten => _NXX,2,Dial(${TRUNKL3}/${EXTEN})
> > > exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
> > >
> > > exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
> > > exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
> > > exten => _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})
> > > exten => _01144XX,1,Dial(SIP/[EMAIL PROTECTED])
> > > exten => _01144XX,2,Dial(${TRUNKL3}/${EXTEN})
> > >
> > > Thanks, David
> > >
> > > --
> > > This message has been scanned for viruses and
> > > dangerous content by KE6UPI, and is
> > > believed to be clean.
> > > KE6UPI thanks MailScanner for their support.
> > > Please contact [EMAIL PROTECTED] if you have
> > > questions about this email.
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/aster> isk-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
>
> --
> This message has been scanned for viruses and
> dangerous content by KE6UPI, and is
> believed to be clean.
> KE6UPI thanks MailScanner for their support.
> Please contact [EMAIL PROTECTED] if you have
> questions about this email.
>


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[Asterisk-Users] Broadvoice problems with outbound calls {Scanned}

[David Shaw]

Hello All, I sign up with $5.99 broadvoice plan. I made in and outbound
calls OK. I upgraded to unlimited world and now I have problems with
outbound calls. I called broadvoice and they said they would get back it
me.

Here are my sip and extension files.


sip.conf

register => XX:[EMAIL PROTECTED]

[broadvoice]
type=friend
username=XX
fromuser=XX
secret=passwd
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes

[bv-in-1]
type=friend
host=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
allow=ulaw

extensions.conf

exten => _NXX,1,Dial(${TRUNKL2}/${EXTEN})
exten => _NXX,2,Dial(${TRUNKL3}/${EXTEN})
exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])

exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})
exten => _01144XX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _01144XX,2,Dial(${TRUNKL3}/${EXTEN})

Thanks, David

-- 
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Re: [Asterisk-Users] Installing ASTERIS@HOME, How to install on text mode same help? {Scanned}

[David Shaw]

When it asked to install type "linux text"  
without the "".
 
But when I installed my [EMAIL PROTECTED] I believed it just 
installed..
 
David

  - Original Message - 
  From: 
  Max 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, February 02, 2005 2:40 
  AM
  Subject: [Asterisk-Users] Installing [EMAIL PROTECTED],How to install on text mode 
  same help? {Scanned}
  
  
  Hello, Thanks for Help!
   
  when try to install [EMAIL PROTECTED] powered by CEntOS
  normal boot, 3 minutes latter:
   
  "You are using unsupported hardware by 
  CentOS,  press OK" if press OK reboot.
   
  I increment mor ram and CPU:
   
  CPU K6II- 500Mhz196Ram HD 20GB 
  Lan cart 10/100MbFax modem genius (Lucent chipset)Fax Modem 
  USR 33.66Sound OnBoard 
  
  Disk 
  Driver 1.44
  
  CD 
  52X
   
  How to install on text mode?
   
   
  regards!
   
  Max Rivera
  Fprm Brazil.-- 
  This message has been scanned for viruses and dangerous content by MailScanner, and is 
  believed to be clean. MailScanner thanks transtec Computers for their support. 
  Plase contact [EMAIL PROTECTED] if you have questions about this 
  email. 
  
  

  ___Asterisk-Users 
  mailing 
  listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:   
  http://lists.digium.com/mailman/listinfo/asterisk-users-- 
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Re: [Asterisk-Users] Record inbound and outbound calls to and fromone number. {Scanned}

[David Shaw]

No, the syntax is not supported. :(  Thats why I asked.. But Monitor is a
better way to go.

David
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, January 31, 2005 9:28 AM
Subject: Re: [Asterisk-Users] Record inbound and outbound calls to and
fromone number. {Scanned}


> David Shaw wrote:
>
> >DAY=`date "+%m-%d-%y_%H:%m"`
> >
> >
> I wasn't aware that this syntax with the ticks is supported in the
configuration files? Is it?
> -Brett
>
>
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> This message has been scanned for viruses and
> dangerous content by KE6UPI, and is
> believed to be clean.
> KE6UPI thanks MailScanner for their support.
> Please contact [EMAIL PROTECTED] if you have
> questions about this email.
>


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[Asterisk-Users] Record inbound and outbound calls to and from one number.

[David Shaw]

Hello All,

I would like to record inbound and outbound calls to and from one
number.

I tried to add lines to my extensions.conf:

DAY=`date "+%m-%d-%y_%H:%m"`

;outbound
exten => 551212,1,Record(${DAY}:gsm)
exten => 551212,2,Dial(${TRUNKL3}/${EXTEN})

;Inbound
[line2]
exten => 551212,1,Record(${DAY}:gsm)
exten => 551212,2,Dial(SIP/101,20)
exten => 551212,3,Hangup



-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Voicemail attachment not being emailed out {Scanned}

[David Shaw]

I lied it did email me an attachment. Check voice-mail entree line. it
has two comas ,, in it.

I rem out the attach=yes in my voicemail.conf file. 
Then added attach=yes at the end of my entree.
101 => {passwd},David,[EMAIL PROTECTED],attach=yes

Works great..
David


On Thu, 2005-01-27 at 18:42 -0500, Jeff R Glassman wrote:
> I am running [EMAIL PROTECTED]
> 
> Voicemail works fine but does not email out the voicemail attachments.  Any
> suggestion?
> ---
> Voicemail.conf
> 
> [general]
> #include vm_general.inc
> #include vm_email.inc
> [default]
> 
> 201 => {password},Jeff G Laptop,[EMAIL PROTECTED],,attach=yes
> -
> Sip.Conf
> 
> [201]
> username=201
> type=friend
> secret={ACCOUT PASSWORD}
> qualify=no
> port=5060
> nat=yes
> mailbox=201
> host=dynamic
> dtmfmode=rfc2833
> context=from-internal
> canreinvite=no
> callerid="Jeff G Laptop" <201>
> 
> 
> 
> 
> Jeff
> 
> ___
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
-- 
David Shaw <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Voicemail attachment not being emailed out {Scanned}

[David Shaw]

I'm new at this too.
In my voicemail.conf under general I have attach=yes.(This works for all
users) I did try removing it and adding to the end of my users voicemail
entries. Run the test and no attachment. But I'm still new.

David


On Thu, 2005-01-27 at 18:42 -0500, Jeff R Glassman wrote:
> I am running [EMAIL PROTECTED]
> 
> Voicemail works fine but does not email out the voicemail attachments.  Any
> suggestion?
> ---
> Voicemail.conf
> 
> [general]
> #include vm_general.inc
> #include vm_email.inc
> [default]
> 
> 201 => {password},Jeff G Laptop,[EMAIL PROTECTED],,attach=yes
> -
> Sip.Conf
> 
> [201]
> username=201
> type=friend
> secret={ACCOUT PASSWORD}
> qualify=no
> port=5060
> nat=yes
> mailbox=201
> host=dynamic
> dtmfmode=rfc2833
> context=from-internal
> canreinvite=no
> callerid="Jeff G Laptop" <201>
> 
> 
> 
> 
> Jeff
> 
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
-- 
David Shaw <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}

[David Shaw]

 => 500,4,Goto(s,6)  ; Return to the start over message.
> 
> exten => 600,1,Playback(demo-echotest); Let them know what's going on
> exten => 600,2,Echo   ; Do the echo test
> exten => 600,3,Playback(demo-echodone); Let them know it's over
> exten => 600,4,Goto(s,6)  ; Start over
> 
> exten => 8500,1,VoicemailMain
> exten => 8500,2,Goto(s,6)
> 
> 
> [default]
> include => demo
> 
> ; I modified stuff from here down...
> 
> exten=_9NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
> exten=_9NXXNXX, 2, dial(SIP/[EMAIL PROTECTED],30)
> exten=_9NXXNXX, 3, congestion()
> exten=_9NXXNXX, 103, busy()
> 
> [sip]
> exten => 1,1,Dial(SIP/xlite1,20,tr)
> exten => 2,1,Dial(SIP/kphone1,20,tr)
> exten => 3,1,Dial(SIP/xlite2,20,tr)
> exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> 
> [incoming]
> exten => 1,1,Dial(SIP/xlite1,20,tr)
> exten => 2,1,Dial(SIP/kphone1,20,tr)
> exten => 3,1,Dial(SIP/xlite2,20,tr)
> exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> 
> [from-broadvoice]
> exten => 1,1,Dial(SIP/xlite1,20,tr)
> exten => 2,1,Dial(SIP/kphone1,20,tr)
> exten => 3,1,Dial(SIP/xlite2,20,tr)
> exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> 
> ---
> 
>   Steve
> Stephen Amadei
> 5114 Harbor Beach Blvd
> Brigantine Beach, NJ 08203
> (609) 703-9649
> 
> 
> 
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-- 
David Shaw <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Am I missing something really basichere?????helpwith Asterisk@home {Scanned} {Scanned} {Scanned}

[David Shaw]

Remember I'm new here too.

You might need to edit /etc/zaptel.conf 
Check fxsks=1-4 I have four X100P cards.
If you have one X100P change it to fxsks=1

I have no idea what AMP configurator is?

David


On Thu, 2005-01-27 at 12:17 -0500, Jeff R Glassman wrote:
> I also edited the Zapata.conf file I did not change the zaptel.conf,  what 
> did you change in it
> 
> Jeff
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of David Shaw
> Sent: Thursday, January 27, 2005 11:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Am I missing something really
> basichere?helpwith [EMAIL PROTECTED] {Scanned} {Scanned}
> 
> 
> I'm running [EMAIL PROTECTED] I had to edit /etc/zaptel.conf
> and /etc/asterisk/zapata.conf. After that it works great. 
> 
> David..
> 
> 
> On Thu, 2005-01-27 at 09:54 -0500, dean collins wrote:
> > Ok, I thought the point of [EMAIL PROTECTED] was that it automatically
> > detected the X100P board and configured it correctly.
> > 
> >  
> > 
> > Is this incorrect? You still need to modify /etc/zaptel files? And not
> > just using the AMP configurator.
> > 
> >  
> > 
> > There is no mention of this on the [EMAIL PROTECTED] webpage.
> > 
> >  
> > 
> > Can anyone who has actually used [EMAIL PROTECTED] confirm this one way
> > or the other?
> > 
> >  
> > 
> >  
> > 
> > Thanks,
> > 
> > Dean
> > 
> >  
> > 
> >  
> > 
> >  
> > 
> >
> > __
> > 
> > From:[EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of David Shaw
> > Sent: Thursday, January 27, 2005 9:28 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Am I missing something really basic
> > here?helpwith [EMAIL PROTECTED] {Scanned}
> > 
> > 
> >  
> > 
> > Yes, You need to add channels to your zapata.conf file.
> > 
> > 
> >  
> > 
> > 
> > zapata.conf
> > 
> > 
> > [channels]
> > ;
> > ; X100P plugged into PSTN
> > ; X100P # 1
> > ;[line1]
> > context=line1
> > signalling=fxs_ks
> > echocancel=yes
> > echocancelwhenbridged=yes
> > relaxdtmf=yes
> > rxgain=1.5
> > txgain=1.5
> > immediate=no
> > busydetect=no
> > callprogress=no
> > musiconhold=default
> > usecallerid=yes
> > callerid=asreceived
> > channel => 1
> > 
> > 
> >  
> > 
> > 
> > You might need to edit /etc/zaptel.conf
> > 
> > 
> > Check fxsks=1-4 I have four X100P cards.
> > 
> > 
> > If you have one change it to fxsks=1
> > 
> > 
> >  
> > 
> > 
> > extensions.conf
> > 
> > 
> >  
> > 
> > 
> > [general]
> > static=yes
> > writeprotect=no
> > 
> > 
> >  
> > 
> > 
> > [globals]
> > CONSOLE=Console/dsp ; Console interface
> > for demo
> > IAXINFO=guest   ; IAXtel
> > username/password
> > TRUNKL1=Zap/1
> > TRUNKL2=Zap/2
> > TRUNKL3=Zap/3
> > TRUNKL4=Zap/4   ; Trunk interface
> > TRUNKMSD=1  ; MSD digits to strip
> > (usually 1 or 0)
> > 
> > 
> >  
> > 
> > 
> > [line1]
> > exten => s,1,Dial(SIP/101,20)
> > exten => s,2,Answer
> > exten => s,3,Wait,1
> > exten => s,4,Voicemail,101
> > exten => s,5,Hangup
> > 
> > 
> >  
> > 
> > 
> > Here I have TRUNKL1=Zap/? for each X100P cards.
> > 
> > 
> >  
> > 
> > 
> > [line1] tells asterisk how to answer that line. 
> > 
> > 
> >  
> > 
> > 
> > Remember I'm very new at this, but I didn't see anyone respond to your
> > post.
> > 
> > 
> >  
> > 
> > 
> > Goog luck, David
> > 
> > 
> >  
> > 
> > 
> >  
> > 
> > 
> >  
> > 
> > 
> >  
> > 
> > 
> > - Original Message - 
> > 
> > 
> > From: dean collins 
> > 
> > 
> > To: A

RE: Re: [Asterisk-Users] Changing mailbox greeting {Scanned}

[David Shaw]

I over looked the "u" or "b" in front of the extension.

Works great. 
Thanks, David


On Thu, 2005-01-27 at 19:44 +0100, Michiel van Baak wrote:
> > The unavailable msg is used when the voicemail app picks up after no phone 
> > on your extension picks up, ie you are not available to answer the phone.
> > 
> > Dan
> > 
> > On Thu, 27 Jan 2005, David Shaw wrote:
> > 
> > > Hello, I have tried to change my mailbox greeting. I go into Mailbox
> > > options > 1 record your unavailable msg.
> > > options > 2 Record your busy msg
> > > options > 3 your name.
> > >
> > > So how to I get it to play the unavailable msg??
> > >
> > > --
> > > David Shaw 
> > >
> 
> To force the voicemail system to play the unavail msg you can also add this 
> in your extensions.conf: exten => 13,3,VoiceMail(u${vmbox13})
> 
> Note the u
> You can also use the b to force to play the busy msg.
> 
> Happy *-ing
> Michiel
> 
> 
> 
> ___
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-- 
David Shaw <[EMAIL PROTECTED]>

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[Asterisk-Users] Changing mailbox greeting

[David Shaw]

Hello, I have tried to change my mailbox greeting. I go into Mailbox
options > 1 record your unavailable msg.
options > 2 Record your busy msg
options > 3 your name. 

So how to I get it to play the unavailable msg??

-- 
David Shaw <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Am I missing something really basic here?????helpwith Asterisk@home {Scanned} {Scanned}

[David Shaw]

I'm running [EMAIL PROTECTED] I had to edit /etc/zaptel.conf
and /etc/asterisk/zapata.conf. After that it works great. 

David..


On Thu, 2005-01-27 at 09:54 -0500, dean collins wrote:
> Ok, I thought the point of [EMAIL PROTECTED] was that it automatically
> detected the X100P board and configured it correctly.
> 
>  
> 
> Is this incorrect? You still need to modify /etc/zaptel files? And not
> just using the AMP configurator.
> 
>  
> 
> There is no mention of this on the [EMAIL PROTECTED] webpage.
> 
>  
> 
> Can anyone who has actually used [EMAIL PROTECTED] confirm this one way
> or the other?
> 
>  
> 
>  
> 
> Thanks,
> 
> Dean
> 
>  
> 
>  
> 
>  
> 
>
> __
> 
> From:[EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of David Shaw
> Sent: Thursday, January 27, 2005 9:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Am I missing something really basic
> here?helpwith [EMAIL PROTECTED] {Scanned}
> 
> 
>  
> 
> Yes, You need to add channels to your zapata.conf file.
> 
> 
>  
> 
> 
> zapata.conf
> 
> 
> [channels]
> ;
> ; X100P plugged into PSTN
> ; X100P # 1
> ;[line1]
> context=line1
> signalling=fxs_ks
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=1.5
> txgain=1.5
> immediate=no
> busydetect=no
> callprogress=no
> musiconhold=default
> usecallerid=yes
> callerid=asreceived
> channel => 1
> 
> 
>  
> 
> 
> You might need to edit /etc/zaptel.conf
> 
> 
> Check fxsks=1-4 I have four X100P cards.
> 
> 
> If you have one change it to fxsks=1
> 
> 
>  
> 
> 
> extensions.conf
> 
> 
>  
> 
> 
> [general]
> static=yes
> writeprotect=no
> 
> 
>  
> 
> 
> [globals]
> CONSOLE=Console/dsp ; Console interface
> for demo
> IAXINFO=guest   ; IAXtel
> username/password
> TRUNKL1=Zap/1
> TRUNKL2=Zap/2
> TRUNKL3=Zap/3
> TRUNKL4=Zap/4   ; Trunk interface
> TRUNKMSD=1  ; MSD digits to strip
> (usually 1 or 0)
> 
> 
>  
> 
> 
> [line1]
> exten => s,1,Dial(SIP/101,20)
> exten => s,2,Answer
> exten => s,3,Wait,1
> exten => s,4,Voicemail,101
> exten => s,5,Hangup
> 
> 
>  
> 
> 
> Here I have TRUNKL1=Zap/? for each X100P cards.
> 
> 
>  
> 
> 
> [line1] tells asterisk how to answer that line. 
> 
> 
>  
> 
> 
> Remember I'm very new at this, but I didn't see anyone respond to your
> post.
> 
> 
>  
> 
> 
> Goog luck, David
> 
> 
>  
> 
> 
>  
> 
> 
>  
> 
> 
>  
> 
> 
> - Original Message - 
> 
> 
> From: dean collins 
> 
> 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> 
> Sent: Wednesday, January 26, 2005 5:36 AM
> 
> 
> Subject: [Asterisk-Users] Am I missing something really basic
> here? helpwith [EMAIL PROTECTED] {Scanned}
> 
> 
>  
> 
> 
> Iâm trying to install [EMAIL PROTECTED], Iâve just downloaded the
> latest cd from soundforge. I can get it to install ok (network
> card didnât auto configure â but I worked out how to use
> ânetconfigâ).
> 
>  
> 
> I worked out how to add a few grandstream budgetone fine.
> Worked out how to upload music etc. Worked out how to modify
> FOP.
> 
>  
> 
> Voicemail and meetmeâs work fine.
> 
>  
> 
> HOWEVERâ.
> 
>  
> 
> Iâm using a X100p. I cant get it to make a call out or use the
> default extension for an incoming line.
> 
>  
> 
> What do I need to make the pstn connection work? Do I need to
> modify Zapata.conf? there are zero instructions on the
> [EMAIL PROTECTED] page as to what to do.
> 
>  
> 
> Can anyone help me out here.
> 
>  
> 
>  
> 
> TIA,
> 
> Dean
> 
> 
> -- 
> This message has been scanned

Re: [Asterisk-Users] Am I missing something really basic here????? helpwith Asterisk@home {Scanned}

[David Shaw]

Yes, You need to add channels to your zapata.conf 
file.
 
zapata.conf
[channels];; X100P plugged into PSTN; 
X100P # 
1;[line1]context=line1signalling=fxs_ksechocancel=yesechocancelwhenbridged=yesrelaxdtmf=yesrxgain=1.5txgain=1.5immediate=nobusydetect=nocallprogress=nomusiconhold=defaultusecallerid=yescallerid=asreceivedchannel 
=> 1
 
You might need to edit 
/etc/zaptel.conf
Check fxsks=1-4 I have four X100P 
cards.
If you have one change it to fxsks=1
 
extensions.conf
 
[general]static=yeswriteprotect=no
 
[globals]CONSOLE=Console/dsp 
; Console interface for 
demoIAXINFO=guest   
; IAXtel 
username/passwordTRUNKL1=Zap/1TRUNKL2=Zap/2TRUNKL3=Zap/3TRUNKL4=Zap/4   
; Trunk 
interfaceTRUNKMSD=1  
; MSD digits to strip (usually 1 or 0)
 
[line1]exten => 
s,1,Dial(SIP/101,20)exten => s,2,Answerexten => 
s,3,Wait,1exten => s,4,Voicemail,101exten => 
s,5,Hangup
 
Here I have TRUNKL1=Zap/? for each X100P 
cards.
 
[line1] tells asterisk how to answer that line. 

 
Remember I'm very new at this, but I didn't see 
anyone respond to your post.
 
Goog luck, David

 
 
 

  - Original Message - 
  From: 
  dean 
  collins 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 26, 2005 5:36 
  AM
  Subject: [Asterisk-Users] Am I missing 
  something really basic here? helpwith [EMAIL PROTECTED] {Scanned}
  
  
  I’m trying to install [EMAIL PROTECTED], I’ve just downloaded the latest 
  cd from soundforge. I can get it to install ok (network card didn’t auto 
  configure – but I worked out how to use 
  ‘netconfig’).
   
  I worked out how to add a few 
  grandstream budgetone fine. Worked out how to upload music etc. Worked out how 
  to modify FOP.
   
  Voicemail and meetme’s work 
  fine.
   
  HOWEVER….
   
  I’m using a X100p. I cant get it 
  to make a call out or use the default extension for an incoming 
  line.
   
  What do I need to make the pstn 
  connection work? Do I need to modify Zapata.conf? there are zero instructions 
  on the [EMAIL PROTECTED] page as to what to do.
   
  Can anyone help me out 
  here.
   
   
  TIA,
  Dean-- 
  This message has been scanned for viruses and dangerous content by MailScanner, and is 
  believed to be clean. MailScanner thanks transtec Computers for their support. 
  Plase contact [EMAIL PROTECTED] if you have questions about this 
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RE: [Asterisk-Users] BroadVoice Or VoicePulse ? {Scanned}

[David Shaw]

I have two Vonage lines and one Lingo line. I would like to drop them
and go with BroadVoice. I would need to have three to four lines from
BV. I should be able to configure Asterisk to handle all the SIP
connections? Right

Thanks, David

PS its a home PBX.



On Tue, 2005-01-25 at 11:39, Jay Milk wrote:
> >From my experience, Broadvoice is good except for customer service.
> VoicePulse I wasn't too impressed with.  Determine what your needs are,
> and see if a per-minute plan would work better for you.  You can have US
> calls for 1.3c/minute incoming and outgoing if you shop around.
> 
> -Original Message-
> From: Manjit Riat [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, January 25, 2005 12:09 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] BroadVoice Or VoicePulse ?
> 
> 
> Which would you recommend as far and quality and pricing to connect to
> asterisk (including DTMF issues)/
> 
> ___
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Re: [Asterisk-Users] One Ring Mystery {Scanned}

[David Shaw]

I could have message waiting ring. It is a default SPA-2000 setup!!
I will check for msg waiting ring tonight and check for msg's.

Thanks, David


- Original Message -
From: "Asterisk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, January 25, 2005 10:27 AM
Subject: Re: [Asterisk-Users] One Ring Mystery {Scanned}


> Are you sure that it's not a "you have a message waiting ring" ?
>
> Julian
>
> David Shaw wrote:
>
> >Out of the blue extension 100 will ring once. This will happen 3-4 times
> >a day. I have checked the logs and no incoming calls. I have extension
> >100 and 101 going to a SPA-2000 ATA. Any ideas on this??
> >
> >Thanks, David
> >
> >___
> >Asterisk-Users mailing list
> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
>
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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> --
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> dangerous content by KE6UPI, and is
> believed to be clean.
> KE6UPI thanks MailScanner for their support.
> Please contact [EMAIL PROTECTED] if you have
> questions about this email.
>


-- 
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Re: [Asterisk-Users] One Ring Mystery {Scanned}

[David Shaw]

Sounds good to me. I should see a new register for the extension in the
logs?

Thanks, David


- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, January 25, 2005 9:50 AM
Subject: Re: [Asterisk-Users] One Ring Mystery {Scanned}


> David Shaw wrote:
> > Out of the blue extension 100 will ring once. This will happen 3-4 times
> > a day. I have checked the logs and no incoming calls. I have extension
> > 100 and 101 going to a SPA-2000 ATA. Any ideas on this??
>
> As far as I can tell that means the phone lost it's connection to
> Asterisk, and then when it gets the connection back it will ring once.
> ___
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> --
> This message has been scanned for viruses and
> dangerous content by KE6UPI, and is
> believed to be clean.
> KE6UPI thanks MailScanner for their support.
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[Asterisk-Users] One Ring Mystery

[David Shaw]

Out of the blue extension 100 will ring once. This will happen 3-4 times
a day. I have checked the logs and no incoming calls. I have extension
100 and 101 going to a SPA-2000 ATA. Any ideas on this??

Thanks, David

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Re: [Asterisk-Users] Dialing Delay {Scanned}

[David Shaw]

Thanks Everyone for the help. I want to change over to Broadvoice and dump
the ATAs from Vonage and Lingo. That should help with dialing delays. (I
hope)

Thanks, David



- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, January 24, 2005 12:26 PM
Subject: Re: [Asterisk-Users] Dialing Delay {Scanned}


> On Mon, 2005-01-24 at 13:47 -0600, Eric Wieling wrote:
> > David Shaw wrote:
> >
> > > exten => 510,1,Dial(SIP/510,20)
> > > exten => 510,2,Voicemail,510
> > >
> > > exten => 8500,1,VoicemailMain
> > >
> > > exten => _NXX,1,Dial(${TRUNKL4}/${EXTEN})
> > > exten => _NXX,2,Dial(${TRUNKL2}/${EXTEN})
> > > exten => _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
> > > exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
> > > exten => _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
> >
> > You have overlapping patterns.  How does Asterisk know that you are
> > dialing "510" and not "510-1234"?  It doesn't.  That's why people use
> > "dial 9" and never start their extensions with "9".
> >
> > Try:
> > exten => _9NXX,1,Dial(${TRUNKL4}/${EXTEN:1})
> > exten => _9NXX,2,Dial(${TRUNKL2}/${EXTEN:1})
> > exten => _91NXXNXX,1,Dial(${TRUNKL4}/${EXTEN:1})
> > exten => _91NXXNXX,2,Dial(${TRUNKL2}/${EXTEN:1})
> > exten => _901144XX,1,Dial(${TRUNKL3}/${EXTEN:1})
> >
> > Also you do not want to just blindly dial the call again in priority
> > 2.  You want to find out what the status was of the previous Dial (See
> > "show application dial", README.variables, and the stdexen macro in
> > the extensions.conf.sample.
>
> You also missed what appears to be another set of problems.
>
> > IAXINFO=guest   ; IAXtel
username/password
> > TRUNKL1=Zap/1   ; Vonage line 1
> > TRUNKL2=Zap/2   ; Vonage line 2
> > TRUNKL3=Zap/3   ; Lingo line 1
> > TRUNKL4=Zap/4   ; Verizon home line
>
> Specifically, it appears that he is using a TDM card to dial an external
> adapter for Vonage. So that means you enter the digits yourself, then
> asterisk enters them on the analog line to the adapter, then it has to
> signal it out to the far side. So there is a possibility that there is
> also a pattern match problem on the vonage SIP device
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] Dialing Delay {Scanned}

[David Shaw]

Here is my extensions.conf. Remember I'm a newbe.

Long delays on any out going call.


[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKL1=Zap/1
TRUNKL2=Zap/2
TRUNKL3=Zap/3
TRUNKL4=Zap/4 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)


[default]
;include => demo
exten => 300,1,Macro(stdexten,1234,SIP/300)
exten => 301,1,Macro(stdexten,1234,SIP/301)

;SpeedDial
exten => 01,1,Dial(${TRUNKL2}/X})
exten => 02,1,Dial(${TRUNKL2}/X})
exten => 03,1,Dial(${TRUNKL2}/X}) ;Mum & Jim
exten => 05,1,Dial(${TRUNKL2}/X}) ;Kevin

exten => 100,1,Dial(SIP/100,30)
exten => 100,2,Voicemail,100

exten => 101,1,Dial(SIP/101,20)
exten => 101,2,Voicemail,101

exten => 102,1,Dial(SIP/102,20)
exten => 102,2,Voicemail,101

exten => 510,1,Dial(SIP/510,20)
exten => 510,2,Voicemail,510

exten => 8500,1,VoicemailMain

exten => _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _01144XX,1,Dial(${TRUNKL3}/${EXTEN})

[line1]
exten => s,1,Dial(SIP/101,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,101
exten => s,5,Hangup

[line2]
exten => s,1,Answer
exten => s,2,Wait,1
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,Authenticate(X)
exten => s,6,DISA,no-password|default

[line3]
exten => s,1,Dial(SIP/510,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,510
exten => s,5,Hangup

[line4]
exten => s,1,Dial(SIP/100,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,100
exten => s,5,Hangup



- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, January 24, 2005 8:32 AM
Subject: Re: [Asterisk-Users] Dialing Delay {Scanned}


> David Shaw wrote:
>  > Hello, When I dial out there is a long delay in dialing.
>
> What are you dialing out to?
> What are you dialing out from?
> What does your config look like for the answer to the above questions?
>
>  > Is this normal?
>
> It varies. The dialing sequence from Stargate Command takes longer than
> the dialing sequence from Atlantis. I believe this is due to the true
> DHD at Atlantis, and notably upgraded Stargate. I am open for discussion.
;)
>
> As you can see, your question was rather vague, and was nowhere near
> specific enough to get the answer you were seeking. Unless, you just
> happenned to be wondering about the Stargate...
>
> --
> Andrew Thompson
> http://aktzero.com/
> http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] Dialing Delay {Scanned}

[David Shaw]

Go easy on me I'm new.

extensions.conf
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNKL1=Zap/1   ; Vonage line 1
TRUNKL2=Zap/2   ; Vonage line 2
TRUNKL3=Zap/3   ; Lingo line 1
TRUNKL4=Zap/4   ; Verizon home line
TRUNKMSD=1

[default]
;include => demo
exten => 300,1,Macro(stdexten,1234,SIP/300)
exten => 301,1,Macro(stdexten,1234,SIP/301)

;SpeedDial
exten => 01,1,Dial(${TRUNKL2}/XXX})
exten => 02,1,Dial(${TRUNKL2}/XXX})
exten => 03,1,Dial(${TRUNKL2}/XXX}) ;Mum & Jim
exten => 05,1,Dial(${TRUNKL2}/XXX}) ;Kevin

exten => 100,1,Dial(SIP/100,30)
exten => 100,2,Voicemail,100

exten => 101,1,Dial(SIP/101,20)
exten => 101,2,Voicemail,101

exten => 102,1,Dial(SIP/102,20)
exten => 102,2,Voicemail,101

exten => 510,1,Dial(SIP/510,20)
exten => 510,2,Voicemail,510

exten => 8500,1,VoicemailMain

exten => _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _01144XX,1,Dial(${TRUNKL3}/${EXTEN})

[line1]
exten => s,1,Dial(SIP/101,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,101
exten => s,5,Hangup

[line2]
;exten => s,1,Dial(SIP/101,20)
exten => s,1,Answer
exten => s,2,Wait,1
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,Authenticate(XX)
exten => s,6,DISA,no-password|default

;[line2
;exten => s,1,Dial(SIP/101,20)
;exten => s,2,Answer
;exten => s,3,Wait,1
;exten => s,4,Voicemail,101
;exten => s,5,Hangup

[line3]
exten => s,1,Dial(SIP/510,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,510
exten => s,5,Hangup

[line4]
exten => s,1,Dial(SIP/100,20)
exten => s,2,Answer
exten => s,3,Wait,1
exten => s,4,Voicemail,100
exten => s,5,Hangup







> On Mon, 2005-01-24 at 07:42 -0800, David Shaw wrote:
>> Hello, When I dial out there is a long delay in dialing. Is this normal?
>
> No it isn't normal.
>
> Examine/post relevant portions of config files and explain what
> interfaces you are using.
>
> Quick guess is the pattern match for your outbound calls is waiting for
> a timeout instead of matching a real specific pattern.
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
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Thanks, David

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[Asterisk-Users] Dialing Delay

[David Shaw]

Hello, When I dial out there is a long delay in dialing. Is this normal?

Thanks,
David

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Re: [Asterisk-Users] Web min Module for Asterisk (and thirdlane) {Scanned}

[David Shaw]

I'm a firefighter and we have a lot time off. So I help out with the other
firefighters. They have small businesses and I install Linux Servers. They
seem to like Webmin. I have one firefighter that runs a small pool clean
business and needs a small PBX.(3 lines) Anyways, its just easyer on me to
show them how to make changes on webmin.

David


> Webmin modules are usually written in perl but there is always the
> possibility that the perl invokes some compiled binaries. I would always
> ask if 100% source is included. Otherwise you can end up paying for
> something that someday prevents you from upgrading the overall linux
> system.
>
> Don't forget that looking at proprietary source code can put you in a
> position where your contributions to open source might be challenged.
> They might falsely accuse you of "borrowing" from the proprietary code
> you saw. It can get complicated. Ask me to work on proprietary software
> for your senseless widget manufacturing plant and I will give you a
> great deal. Ask me to work on proprietary software for something I am
> really interested in like voip and I will ask you to compensate me for
> every day that the non-disclosure and do-not-compete clauses are in
> effect. I guess I'm poor but free.
>
> I have thought about writing a free * webmin module. It would definitely
> have to use the * remote management features. Reason for that is maybe I
> don't want webmin(and many other things) running on a busy * server. You
> can always have webmin available on the server for maintainance needs
> but not started at boot time. The rest of the time you use webmin and
> the * module on another machine to manage various * servers. You can use
> stunnel to encrypt management traffic to remote servers. Take a really
> good look at Jamie Cameron's webmin and you will see that it was
> designed with this type of need in mind. You could use a single webmin
> server to manage a building with dozens of networked servers. He already
> put lots of tools in the toolbox for the module developer.
>
> Anyway, I would agree with Henry for my own situation. But maybe you
> would quickly get your $300 worth because you want to have a
> cross-trained pbx management team. Or maybe you can get a good price on
> this and bundle it with systems you sell. The screen shots on the
> website didn't tell me enough to make me salivate over it.
>
> --
> Paul
>
> Henry Devito wrote:
>
>>
>>
>>
>>
>>>-Original Message-
>>>From: [EMAIL PROTECTED]
>>>
>>>
>>[mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of
>>[EMAIL PROTECTED]
>>
>>
>>>Sent: Friday, January 21, 2005 9:28 AM
>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane)
>>>
>>>Henry Devito wrote:
>>>
>>>
>>>

www.thirdlane.com   has already written a
close dsource webmin module.  I have no idea how much it costs or how
well it works.





>>>I've attempted to contact thirdlane to get pricing on their GUI and
>>>can't seem to get anyone to reply.
>>>
>>>My personal feeling is that if it's closed source, the support better be
>>>excellent. And if I can't get a reply to a sales question.. What's going
>>>to happen when I have a problem?
>>>
>>>Ek!
>>>-Brett
>>>
>>>
>>
>>I could not agree more.  Last time a new the gui was $300 US.  I emailed
>> a
>>sales person about 2 months ago and that was his response.  I never
>>purchased it for the simple reason that I didn't need it.  Once you learn
>>the command line it seems just as easy to edit config files that way.
>>
>>Have a good day
>>Henry
>>
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Thanks, David

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Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane) {Scanned}

[David Shaw]

Just got a reply from them. $300 Commercial $100 personal. They gave me
a login to look at the demo. Didn't see anything I wanted to buy.

David

On Fri, 2005-01-21 at 09:34, C F wrote:
> Interesting, because I also called them, and I was able to get a
> price, they told me $300 per license, for bulk every 5th license is
> free.
> 
> 
> On Fri, 21 Jan 2005 11:43:13 -0500, Ferguson, Michael
> <[EMAIL PROTECTED]> wrote:
> > Same here.
> > I called them yesterday plus email and still no reply.
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: Friday, January 21, 2005 10:28 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane)
> > 
> > Henry Devito wrote:
> > 
> > >
> > > www.thirdlane.com   has already written a
> > > close dsource webmin module.  I have no idea how much it costs or how
> > > well it works.
> > >
> > >
> > >
> > I've attempted to contact thirdlane to get pricing on their GUI and
> > can't seem to get anyone to reply.
> > 
> > My personal feeling is that if it's closed source, the support better be
> > 
> > excellent. And if I can't get a reply to a sales question.. What's going
> > 
> > to happen when I have a problem?
> > 
> > Ek!
> > -Brett
> > 
> > ___
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Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane) {Scanned}

[David Shaw]

I'm playing with [EMAIL PROTECTED] and it has a web interface. Great for
users to check Voicemail. Anyways I would help with webmin but I have
never wrote html before and I'm very new to Asterisk.

Thanks, David

On Fri, 2005-01-21 at 07:27, [EMAIL PROTECTED] wrote:
> Henry Devito wrote:
> 
> >  
> > www.thirdlane.com   has already written a 
> > close dsource webmin module.  I have no idea how much it costs or how 
> > well it works.
> >  
> >  
> >
> I've attempted to contact thirdlane to get pricing on their GUI and 
> can't seem to get anyone to reply.
> 
> My personal feeling is that if it's closed source, the support better be 
> excellent. And if I can't get a reply to a sales question.. What's going 
> to happen when I have a problem?
> 
> Ek!
> -Brett
> 
> 
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Re: [Asterisk-Users] Help Me please {Scanned}

[David Shaw]

I'm new to this as well. I use SJPhone as my software. Works good. The
default install of Asterisk has allot extensions and voice mail boxes to
play with.

http://google.com  asterisk sjphone and you will find allot of good
stuff.

David


On Fri, 2005-01-21 at 05:20, Krishnan wrote:
> I made one sever with the XORCOM RAPID
> will you tell me how can i use it with softphones? and
> from where i can download that
> actually am a newbie on this. so please give me help
> this is to work on a lan. so how can i do that
> hope that you can give me a clear help or clear link
>  with regards
>   Krishnan
> 
> 
> 
>   
>   
>   
> ___ 
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Re: [Asterisk-Users] Troubles with Broadvoice (register) {Scanned}

[David Shaw]

Thinking out load here. 

Could we replace the hostname in the sip.conf to something like
"broadvoice". Then have a script ping all the proxys for broadvoice then
write the best IP address in the /etc/hosts?

sip.conf
register => @broadvoice

host file
XXX.XXX.XXX.XXX broadvoice

This should keep us from reloading asterisk

David

Just thinking out load

On Thu, 2005-01-20 at 11:11, Helder RogÃrio [MICROREDE] wrote:
> It was a problem regarding the register => not being at the top of the
> [general] section of sip.conf
> 
> - Original Message - 
> From: "Helder RogÃrio [MICROREDE]" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Thursday, January 20, 2005 4:21 PM
> Subject: Re: [Asterisk-Users] Troubles with Broadvoice (register)
> 
> 
> > Hi!
> >
> > But the only server they gave for sip registration is sip.broadvoice.com I
> > have several for outbound proxy proxy.chi.broadvoice.com and etc...
> >
> > Do you have any other for sip?
> >
> > Best regards,
> > Helder
> >
> >
> > - Original Message - 
> > From: "Paul" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Thursday, January 20, 2005 4:15 PM
> > Subject: Re: [Asterisk-Users] Troubles with Broadvoice (register)
> >
> >
> > > Sometimes I have problems and changing to another of their servers makes
> > > it start working again. There probably is a way to make * deal with this
> > > properly. I am using the broadvoice account for test purposes at this
> > > time so I just edit sip.conf and restart * when this happens. What I
> > > have observed is that the server I can't register with will still have
> > > good ping times when this happens.
> > >
> > > Helder RogÃrio [MICROREDE] wrote:
> > >
> > > >Hi!
> > > >
> > > >Are you also getting in trouble while trying to register in Broadvoice?
> > > >
> > > >Cumprimentos / Best regards,
> > > >
> > > >Helder RogÃrio
> > > >
> > > >
> > > >__
> > > >Microrede - Tecnologias de InformaÃÃo, Ltd.
> > > >http://www.microrede.pt
> > > >
> > > >***
> > > > Â There are only two types of people in the world, those who have lost
> > data
> > > >and those who will. Â
> > > >-- Richard Nixon
> > > >
> > > >___
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> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
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> > > >
> > > >
> > >
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Re: [Asterisk-Users] RE: E911 Testing ! {Scanned}

[David Shaw]

As a Firefighter I would call the 911 Dispatch Center first using there
office number (XXX-). Tell them what you would like to do, what
time, address and phone number. Then give them you cell. If something
happens with the test they will call you back on your cell. Also some
Dispatch centers can run the same test on one of there business lines.

David


On Wed, 2005-01-19 at 14:21, Jason Kawakami wrote:
> -Original Message-
> 
> I believe the 911 is a serious issue if one does an asterisk installation in
> an office. How do you test 911? Won't they arrest you or something for
> dialing 911 for no reason and talking to one of their agents who could have
> taken a more important call?
> 
> -speaking from 10+ years of installations, dial 911 and tell the operator
> your name, who you are with, and that you are testing a new phone system.
> Confirm with them that the telephone number and address they have in their
> system is correct, say thank you and hang-up.  
> 
> On occasion, you get a surly operator who has had a bad day but crap, if you
> had their job, your days may not be so good either.
>  
> 
> On the other hand what an emergency comes up (like someone got seriously
> injured) and on top of that asterisk crashed all of a sudden bringing the
> whole office PBX down. Since it would be not be possible to place a call and
> emergency matter becomes more serious, who would be held responsible? The
> person who installed the PBX for not implementing a redundant and reliable
> system?
> 
> -document that on 'X' date and 'Y' time, you tested and confirmed that 911
> access was functioning and have the client sign off on the installation.
> After that, the system is theirs.  
> 
> Always test emergency services access for premises equipment based solutions
> unless you have signed documentation from the client that they do not want
> 911 access out of their system!
> 
> Jason Kawakami
> www.optellabs.com
> Salt Lake City, UT
> 
> 
> 
> 
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Re: [Asterisk-Users] Webmin Module for Asterisk {Scanned}

[David Shaw]

I will try out your pages..

Thanks, David

PS I would love to work on your Asterisk Webmin pages, but I don't know
how.


On Thu, 2005-01-20 at 06:01, [EMAIL PROTECTED] wrote:
> There is already one, you can find it here :
> ftp://ftp.asterisk.org/pub/asterisk/webmin
> 
> But I never managed to make it work, maybe it should be updated
> 
> Anybody wanna take the challenge ? :)
> 
> BTW, I've done some web pages that show you your configuration, and
> let you edit the text files in your browser. If you want it, drop me a
> message
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[Asterisk-Users] Webmin Module for Asterisk

[David Shaw]

Is there any hope on a Webmin Module for Asterisk?

Thanks, David

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Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}

[David Shaw]

I had signed up with BV and wasn't able to call in or out. I had called BV
support and said they wouldn't help if I didn't install the patch. They did
say I was connected to them. Basically I had errors installing the patch so
I asked for a refund. I'm still very new to Asterisk. I did order the
Asterisk book

I guess in the long run I don't need the patch. I'm using the latest CVS
download from Tuesday.

Thanks, David

- Original Message -
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, January 19, 2005 1:53 AM
Subject: Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}


> FYI, that patch was only needed by those that ran * behind a nat/fw box,
> and its primary purpose was to reduce registration traffic at broadvoice.
> Technically, the patch isn't needed to make * work with bv. (The issue
> was that bv would disable an * userid/password if they found your
> system to be re-registering frequently.)
>
> > i use broadvoice with asterisk with no issues it works, and i didnt
> > need the patch
> >
> > On Tue, 2005-01-18 at 19:46 -0800, David Shaw wrote:
> > > Thanks Paul, This is over my head for now. I will sign up for
broadvoice
> > > next week and see if it works. I had signed up for Broadvoice last
week
> > > had problems so I drop them. They said I had connected but I couldn't
> > > receive or place calls with them.
> > >
> > > Thanks, David
> > >
> > >
> > >
> > > > It seems highly likely that it would be in cvs by now. I would not
use
> > > > the file produced by this patch run without first finding out why
hunk
> > > > #12 succeeded. One possible cause would be that the original hunk
has
> > > > been reversed in cvs for some good reason. Note that I usually use
this
> > > > safe and sane test when working with modified sources from debian or
> > > > another distro. Sometimes the package maintainers apply some patches
to
> > > > the upstream source but they don't always clearly document that.
Anyway,
> > > > I used this method to patch and build * for debian. It was not the
cvs
> > > > version but it did the job(see if broadvoice will work). I will
probably
> > > > create a debian package set from cvs next but I need to have a 1.03
> > > > machine running alongside the cvs machine in order to file truly
> > > > meaningful bug reports :)
> > > >
> > > > Dalon Westergreen wrote:
> > > >
> > > >>Hi all,
> > > >>
> > > >>http://edvina.net/broadvoice/patch.shtml
> > > >>
> > > >>this site claims that the patch was incorporated into the * cvs tree
> > > >>as of 12/12/2004
> > > >>
> > > >>--dalon
> > > >>
> > > >>On Tue, 18 Jan 2005 13:15:01 -0800, David Shaw <[EMAIL PROTECTED]>
> > > >> wrote:
> > > >>
> > > >>
> > > >>>I ran a test patch like Paul said.
> > > >>>
> > > >>>[EMAIL PROTECTED] test]# patch chan_sip.c broadvoicesip2.txt
> > > >>>patching file chan_sip.c
> > > >>>Hunk #1 FAILED at 231.
> > > >>>Hunk #2 FAILED at 323.
> > > >>>Hunk #3 FAILED at 494.
> > > >>>Hunk #4 FAILED at 502.
> > > >>>Hunk #5 FAILED at 3738.
> > > >>>Hunk #6 FAILED at 4049.
> > > >>>Hunk #7 FAILED at 4071.
> > > >>>Hunk #8 FAILED at 4091.
> > > >>>Hunk #9 FAILED at 4123.
> > > >>>Hunk #10 FAILED at 4165.
> > > >>>Hunk #11 FAILED at 4181.
> > > >>>Hunk #12 succeeded at 4309 (offset 105 lines).
> > > >>>Hunk #13 FAILED at 4320.
> > > >>>Hunk #14 FAILED at 4335.
> > > >>>Hunk #15 FAILED at 4450.
> > > >>>Hunk #16 FAILED at 6402.
> > > >>>Hunk #17 FAILED at 6526.
> > > >>>Hunk #18 FAILED at 6939.
> > > >>>Hunk #19 FAILED at 6974.
> > > >>>18 out of 19 hunks FAILED -- saving rejects to file chan_sip.c.rej
> > > >>>[EMAIL PROTECTED] test]#
> > > >>>
> > > >>>I'm guessing I have a patched verison of Asterisk?
> > > >>>I tested this on a fresh CVS install.
> > > >>>
> > > >>>Thanks, David
> > > >>>
> > > >>&g

Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}

[David Shaw]

Thanks Paul, This is over my head for now. I will sign up for broadvoice
next week and see if it works. I had signed up for Broadvoice last week
had problems so I drop them. They said I had connected but I couldn't
receive or place calls with them.

Thanks, David



> It seems highly likely that it would be in cvs by now. I would not use
> the file produced by this patch run without first finding out why hunk
> #12 succeeded. One possible cause would be that the original hunk has
> been reversed in cvs for some good reason. Note that I usually use this
> safe and sane test when working with modified sources from debian or
> another distro. Sometimes the package maintainers apply some patches to
> the upstream source but they don't always clearly document that. Anyway,
> I used this method to patch and build * for debian. It was not the cvs
> version but it did the job(see if broadvoice will work). I will probably
> create a debian package set from cvs next but I need to have a 1.03
> machine running alongside the cvs machine in order to file truly
> meaningful bug reports :)
>
> Dalon Westergreen wrote:
>
>>Hi all,
>>
>>http://edvina.net/broadvoice/patch.shtml
>>
>>this site claims that the patch was incorporated into the * cvs tree
>>as of 12/12/2004
>>
>>--dalon
>>
>>On Tue, 18 Jan 2005 13:15:01 -0800, David Shaw <[EMAIL PROTECTED]>
>> wrote:
>>
>>
>>>I ran a test patch like Paul said.
>>>
>>>[EMAIL PROTECTED] test]# patch chan_sip.c broadvoicesip2.txt
>>>patching file chan_sip.c
>>>Hunk #1 FAILED at 231.
>>>Hunk #2 FAILED at 323.
>>>Hunk #3 FAILED at 494.
>>>Hunk #4 FAILED at 502.
>>>Hunk #5 FAILED at 3738.
>>>Hunk #6 FAILED at 4049.
>>>Hunk #7 FAILED at 4071.
>>>Hunk #8 FAILED at 4091.
>>>Hunk #9 FAILED at 4123.
>>>Hunk #10 FAILED at 4165.
>>>Hunk #11 FAILED at 4181.
>>>Hunk #12 succeeded at 4309 (offset 105 lines).
>>>Hunk #13 FAILED at 4320.
>>>Hunk #14 FAILED at 4335.
>>>Hunk #15 FAILED at 4450.
>>>Hunk #16 FAILED at 6402.
>>>Hunk #17 FAILED at 6526.
>>>Hunk #18 FAILED at 6939.
>>>Hunk #19 FAILED at 6974.
>>>18 out of 19 hunks FAILED -- saving rejects to file chan_sip.c.rej
>>>[EMAIL PROTECTED] test]#
>>>
>>>I'm guessing I have a patched verison of Asterisk?
>>>I tested this on a fresh CVS install.
>>>
>>>Thanks, David
>>>
>>>- Original Message -
>>>From: "Paul" <[EMAIL PROTECTED]>
>>>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>
>>>Sent: Tuesday, January 18, 2005 10:31 AM
>>>Subject: Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}
>>>
>>>
>>>
>>>>There is a sane and safe way to do this when you are not sure if the
>>>>patches are already in:
>>>>
>>>>1 - create a new "scratch" directory
>>>>
>>>>2 - copy the patch and chan_sip.c to that directory
>>>>
>>>>3 - patch chan_sip.c broadvoicesip2.txt
>>>>
>>>>If you have the right(unpatched) version of the c source you will only
>>>>get messages about hunks suceeding.
>>>>
>>>>ls -al should now yield something like this:
>>>>
>>>>drwxr-xr-x   2 sysadmin sysadmin   4096 2005-01-13 00:42 .
>>>>drwxr-xr-x  16 sysadmin sysadmin   4096 2005-01-17 23:38 ..
>>>>-rw-r--r--   1 sysadmin sysadmin  15116 2004-12-12 06:02
>>>>
>>>>
>>>broadvoicesip2.txt
>>>
>>>
>>>>-rw-r--r--   1 sysadmin sysadmin 291894 2005-01-13 00:42 chan_sip.c
>>>>-rw-r--r--   1 sysadmin sysadmin 287963 2004-10-25 13:57
>>>> chan_sip.c.orig
>>>>
>>>>David Shaw wrote:
>>>>
>>>>
>>>>
>>>>>Hello, I'm trying to patch Asterisk for uses wth BroadVoice. I'm
>>>>>running [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>.
>>>>>
>>>>>Here is the Error:
>>>>>[EMAIL PROTECTED] asterisk]# patch < broadvoicesip2.txt
>>>>>can't find file to patch at input line 8
>>>>>Perhaps you should have used the -p or --strip option?
>>>>>The text leading up to this was:
>>>>>--
>>>>>|Index: channels/chan_sip.c
>>>>>|===
>>>>>|RC

Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}

[David Shaw]

I ran a test patch like Paul said.

[EMAIL PROTECTED] test]# patch chan_sip.c broadvoicesip2.txt
patching file chan_sip.c
Hunk #1 FAILED at 231.
Hunk #2 FAILED at 323.
Hunk #3 FAILED at 494.
Hunk #4 FAILED at 502.
Hunk #5 FAILED at 3738.
Hunk #6 FAILED at 4049.
Hunk #7 FAILED at 4071.
Hunk #8 FAILED at 4091.
Hunk #9 FAILED at 4123.
Hunk #10 FAILED at 4165.
Hunk #11 FAILED at 4181.
Hunk #12 succeeded at 4309 (offset 105 lines).
Hunk #13 FAILED at 4320.
Hunk #14 FAILED at 4335.
Hunk #15 FAILED at 4450.
Hunk #16 FAILED at 6402.
Hunk #17 FAILED at 6526.
Hunk #18 FAILED at 6939.
Hunk #19 FAILED at 6974.
18 out of 19 hunks FAILED -- saving rejects to file chan_sip.c.rej
[EMAIL PROTECTED] test]#

I'm guessing I have a patched verison of Asterisk?
I tested this on a fresh CVS install.

Thanks, David

- Original Message -
From: "Paul" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, January 18, 2005 10:31 AM
Subject: Re: [Asterisk-Users] Broadvoice Patch Error {Scanned}


> There is a sane and safe way to do this when you are not sure if the
> patches are already in:
>
> 1 - create a new "scratch" directory
>
> 2 - copy the patch and chan_sip.c to that directory
>
> 3 - patch chan_sip.c broadvoicesip2.txt
>
> If you have the right(unpatched) version of the c source you will only
> get messages about hunks suceeding.
>
> ls -al should now yield something like this:
>
> drwxr-xr-x   2 sysadmin sysadmin   4096 2005-01-13 00:42 .
> drwxr-xr-x  16 sysadmin sysadmin   4096 2005-01-17 23:38 ..
> -rw-r--r--   1 sysadmin sysadmin  15116 2004-12-12 06:02
broadvoicesip2.txt
> -rw-r--r--   1 sysadmin sysadmin 291894 2005-01-13 00:42 chan_sip.c
> -rw-r--r--   1 sysadmin sysadmin 287963 2004-10-25 13:57 chan_sip.c.orig
>
> David Shaw wrote:
>
> > Hello, I'm trying to patch Asterisk for uses wth BroadVoice. I'm
> > running [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>.
> >
> > Here is the Error:
> > [EMAIL PROTECTED] asterisk]# patch < broadvoicesip2.txt
> > can't find file to patch at input line 8
> > Perhaps you should have used the -p or --strip option?
> > The text leading up to this was:
> > --
> > |Index: channels/chan_sip.c
> > |===
> > |RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
> > |retrieving revision 1.510.2.23
> > |diff -u -r1.510.2.23 chan_sip.c
> > |--- channels/chan_sip.c26 Nov 2004 01:54:11 -
1.510.2.23
> > |+++ channels/chan_sip.c12 Dec 2004 10:57:45 -
> > --
> > File to patch:
> > [EMAIL PROTECTED] asterisk]#
> >
> > What file should I patch???
> >
> > Thanks, David
> >
> >
> > --
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> >
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[Asterisk-Users] Broadvoice Patch Error {Scanned}

[David Shaw]

Hello, I'm trying to patch Asterisk for uses wth 
BroadVoice. I'm running [EMAIL PROTECTED].
 
Here is the Error:
[EMAIL PROTECTED] asterisk]# patch < 
broadvoicesip2.txtcan't find file to patch at input line 8Perhaps you 
should have used the -p or --strip option?The text leading up to this 
was:--|Index: 
channels/chan_sip.c|===|RCS 
file: /usr/cvsroot/asterisk/channels/chan_sip.c,v|retrieving revision 
1.510.2.23|diff -u -r1.510.2.23 chan_sip.c|--- 
channels/chan_sip.c    26 Nov 2004 
01:54:11 -  1.510.2.23|+++ 
channels/chan_sip.c    12 Dec 2004 
10:57:45 ---File to patch:
[EMAIL PROTECTED] asterisk]# 
 
What file should I patch???
 
Thanks, David
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[Asterisk-Users] Looking for a VoIP provider for my Asterisk box. {Scanned}

[David Shaw]

Hello All, I have Vonage and Lingo and like the service, but I would like
to drop there ATA equipment. I tried BroadVoice had them for less then
24hrs.

Anyways I would like to connect Asterisk directly to a VoIP provider
without the use of there ATA equipment.

Thanks, David

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[Asterisk-Users] Not hanging up. {Scanned}

[David Shaw]

Hello All,

After an incoming call goes to voicemail it doesn't hangup.

Extensions Conf file

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

[default]
include => demo
exten => 300,1,Macro(stdexten,1234,SIP/300)
exten => 301,1,Macro(stdexten,1234,SIP/301)

exten => s,1,Dial(SIP/300,10)
exten => s,2,Dial(SIP/301,10)
exten => s,3,Dial,ZAP/2/XXX,10
exten => s,4,Voicemail,1234
exten => s,5,Hangup

exten => _XXX,1,Dial,ZAP/1/${EXTEN}
exten =>  _1XX,1,Dial,ZAP/2/${EXTEN}


Thanks, David

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[Asterisk-Users] Route incoming call on 4 X100P to different Ext. {Scanned}

[David Shaw]

Hello All,

I have 4 X100P cards. I was hoping to have card (line) go to separate ext.

i.e.
Card 1 (XXX)555-0001 My Ext
Card 2 (XXX)555-0002 Wife's Ext
Card 3 (XXX)555-0003 Fax Ext
Card 4 (XXX)555-0004 My and Wife Ext.

This is what I have now and all incoming line rings this one extension.
exten => s,1,Dial(SIP/300,10)

So what is "s" .

Thanks, David


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[Asterisk-Users] Newbe Can't dial local numbers.

[David Shaw]

Hello All,

I loaded [EMAIL PROTECTED] I'm using SLPhones and can connect to mailboxs
on the system. I have one X100P card. I try to dial out but get
rejected.

Any help...

Thanks, David

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[Asterisk-Users] test

[David Shaw]

test

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Re: [Asterisk-Users] Linksys RT31P2 {Scanned}

[David Shaw]

Check this out.

http://voip.weblogsinc.com/entry/0142584371536804/

David

On Fri, 2005-01-07 at 09:15, Richard Cook wrote:
> Hello,
> 
> Is there any way to unlock the Linksys router?
> 
> --
> Richard Cook
> [EMAIL PROTECTED]
> Tel: 705-497-9320  ext 2010
>  
> -- 
> This message has been scanned for viruses and 
> dangerous content by MailScanner, and is 
> believed to be clean. 
> MailScanner thanks transtec Computers for their support. 
> Plase contact [EMAIL PROTECTED] if you have questions about
> this email. 
> 
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