I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup i
> Stephen, What exactly are you trying to accomplish? If you want basic
> call
> in/out you're just about there. Changes need to be made in your
> extensions.conf. Your phones, by default, are in the [default] context.
> In
> other words when making a call it looks for extensions here. To allow
>
> Are you dialing a 1 before every number? That is required unless you make
> another pattern match.
> exten => _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
> Then it becomes 10-digit dialing without the need to dial a 1. If that
> doesn't work open up the asterisk console and attempt to make a call a
> The voicemail command should be Voicemail([EMAIL PROTECTED]) so in
> extensions.conf
> exten => 101,n,Voicemail([EMAIL PROTECTED])
> As for the console when you launch it add v's to set the debugging level
> 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I
> believe. Ju
> Well, after very quickly making a test call it's not Vitelity. It could be
> something with your account? Might want to try opening a support ticket. If
> you want, create a sub account and e-mail me off list the username and
> password and I'll test it with my box or vice versa.
I am now able t
t; _1NXXNXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten => _NXX,1,Set(CALLERID(num)=9045622082)
exten => _NXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _NXX,n,Dial(SIP/[EMAIL PROTECTED])
exten =>
)=XX)
> ;exten => _1NXXNXX,2,Set(CALLERID(name)=XXXXXX)
>
> exten => _1NXXNXX,1,Set(CALLERID(num)=9045622082)
> exten => _1NXXNXX,n,Set(CALLERID(name)="Stephen Reese")
> exten => _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
>
> exten =&
> Any reason not to ring both at once?
> exten => 101,1,Dial(SIP/101&SIP/[EMAIL PROTECTED],20)
> -Darren
That would also work but what if my sip/101 device (softphone) isn't connected.
Currently if my softphone is not connected then the line will go
straight to voicemail. If I remove the voicemai
num)=XX)
;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XX)
exten => _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten => _1NXXNXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten => _NXX,1,Set(CALLERID(num
proxy_register: 1
messages_uri: "100"
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
sntp_server:10.10.10.1
time_zone: EST
dial_template: DIALPLAN
nat_enable: 1
nat_address: 172.16.2.1
nat_received_processing: 1
outbound_proxy_port: 5060
outbond_proxy: n
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese <[EMAIL PROTECTED]> wrote:
> I've searched around and found a few similar situations where the
> phone will call out when using a Asterisk server but not receive
> inbound calls. My issue is a little stranger. If I call out from
> As a last resort (if qualify doesn't help), you could enter this
> (global) to increase the timeout on UDP translations:
> ip nat translation udp-timeout 300 (or greater if you prefer)
>
> It is likely a NAT timeout issue. When you call outbound, you
> 'reactivate' the SIP session in your NAT dev
udp idle-time 900
>
> -Original Message-----
> From: Stephen Reese [mailto:[EMAIL PROTECTED]
> Sent: Saturday, October 18, 2008 14:41
> To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl
> Dunkin
> Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
>
My latency is kind of high and the voice delay is noticeable.
The Asterisk server is on a dedicated host outside of the network. I
am performing PAT/NAT using a Cisco router.
ns1*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese
> Does the latency remain more or less the same regardless of the
> bandwidth load on the pipe?
>
> If so, TOS bits (what you refer to as QoS) won't help you. You've
> either got network issues (very likely if you have an intra-network ping
> of 30 ms) or the outside host you're sending the traffi
> Alex is correct. Always check thereare no half-duplex links in your
> path. If you have an older dsl/cable modem or router that only has a
> 10M ethernet, it is probably half. Also make certain there are no hubs
> in the path. Keep in mind that colissions ar NORMAl for a hlaf duplex
> connection.
lt,101,1)
[outgoing]
; The following gives an Unknown Caller ID
;exten => _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten => _1NXXNXX,2,Set(CALLERID(name)=XX)
exten => _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten => _1NXXNXX,n,Set(CALLERID(name)="Stephen
On Sun, Oct 19, 2008 at 3:55 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> What is vitel-outbound?? an IP address??
> And what version of Asterisk is this?
> Regards,
> Juan
vitel-outbond is the connection to my sip provider
Version 1.6 of Asterisk
I'm able to make incoming and outgoing calls j
On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> First, I think is better to to have SIP/vitel-outbound/${EXTEN} than
> having SIP/[EMAIL PROTECTED]
> And try issuing SIP SET DEBUG on the cli to see what happens when making the
> call, post back what you see making calls
On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> Try reinstalling Asterisk, because in the channel.c this error is returned
> if the channels TEC (in this case SIP) is not found.
> Weird!!
> Let me know if it works.
> Regards,
> Juan
So the extensions.conf and sip.conf
On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> Stephen:
> Your configuration files looks fine. Try from the CLI issuing "originate
> SIP/101 extension [EMAIL PROTECTED]", having the 101 online, then do that with
> "originate SIP/102 extension [EMAIL PROTECTED]". See w
On Mon, Oct 20, 2008 at 12:25 AM, Eric ManxPower Wieling
<[EMAIL PROTECTED]> wrote:
> ast_request: No channel type registered for ''SIP'
>
> Notice the extra ' in the message.
>
> That is either an error in the error message or you have a an extra ' in
> your Dial line. Something like Dial('SI
On Mon, Oct 20, 2008 at 12:23 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> The second call its OK, so the problem it is just with the Dial(SIP/102), so
> try:
> originate SIP/102 application Dial SIP/102
> and
> originate SIP/101 application Dial SIP/102
> and
> originate SIP/102 application Dia
On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> I do not think NAT is the problem, NAT normally gives you problems like one
> way audio or no registration.
> Try calling the SIP/102 on other extension:
> ;TEST
> exten => 1002,1,Dial(SIP,102|20)
> exten => 1002,n,Hangup
l(SIP/102,20)
exten => 102,n,Hangup
exten => 102,n,Voicemail([EMAIL PROTECTED])
exten=>*98,1,VoiceMailMain([EMAIL PROTECTED])
include => inbound
include => outgoing
[inbound]
exten => 9045622082,1,Goto(default,101,1)
[outgoing]
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=90456
On Tue, Oct 21, 2008 at 9:56 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> Try changing:
> exten => 101,1,Dial(SIP/101/20)
> to
> exten => 101,1,Dial(SIP/101|20) or exten => 101,1,Dial(SIP/101,20)
>
> because exten => 101,1,Dial(SIP/101/20) means you are trying to contact ext.
> 20 on through a t
I also tried downgrading to version 1.4-current but that didn't help.
> Oh, typo, but that still didn't cure it
>
> Successful call from from 101 to 102
>
> == Using SIP RTP CoS mark 5
>-- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-08220318",
> "SIP/102,20") in new stack
> == Using SIP
> I also tried downgrading to version 1.4-current but that didn't help.
>
Any other ideas? I'm at a loss.
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On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> What kind of phone are you trying to connect to 101??? and from where?
>
Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can
contact 102 by dialing 101 but not the other way around, I just get a
busy tone.
__
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> And this phone are connected in a local LAN??
> Because I see Asterisk receiving a "Bad request" from 68.156.63.118
> If those phones are not in your local LAN, try with a soft phone first.
> Could be Zoiper or Xlite.
>
2,1,Dial(SIP/102,20)
exten => 102,n,Hangup
exten => 102,n,Voicemail([EMAIL PROTECTED])
Both extensions can call each other and both extensions ring when the
main line is called... Strange but whatever.
On Thu, Oct 23, 2008 at 1:47 PM, Stephen Reese <[EMAIL PROTECTED]> wrote:
> On T
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here: http://www.pasewaldt.com/cme/cme_index.htm
Would anyone like to comment on their experiences using CME with Asterisk
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons <[EMAIL PROTECTED]> wrote:
> Dare I ask why you want to do this?
>
> Dave
I know it seems counter intuitive but I've several examples of it
being done and for me it would be for the experience of working with
CME. A lot of companies utilize Cisco har
x27;s a different story. And I dare
> say that it does sound like a fun project to take on.
>
> Dave
>
> -Original Message-
> From: Stephen Reese [mailto:[EMAIL PROTECTED]
> Sent: Thursday, October 23, 2008 11:53 PM
> To: Asterisk Users Mailing List - Non-Commerc
On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby wrote:
> That typically means you've got an error in your phone specific config file,
> the SEP[MAC].cnf.xml.
>
> You need to login to the phone via ssh and use the log/log login. Once
> you've done that, look at the logs and see what line of the conf
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby wrote:
> I think your featureLabel definition is wrong.
>
> On the login issue, ssh to the ip of the phone and login first with
> the user/pass you defined in the file (admin/123), then at the second
> login prompt use log/log. That should get you the
On Tue, Nov 10, 2009 at 10:13 AM, Warren Selby wrote:
> In your sip.conf file, be sure to specify nat=no for the phone, even
> though the phone is behind a nat device. The cisco phones handle sip
> packets differently than the way asterisk expects, so you have to do
> this in order to make asteris
This is possible as I was just able to get the latest SIP firmware
loaded on my 7942. Make sure to follow the guide using the 7941 as the
SIP firmware differs from the 79x0 versions. Here's two links to help:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+
On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wrote:
> The 7960 and 79x2 use different sip firmwares and as far a I have seen
> the 7960 does not have the same port issue the 7941/2 seems to have
> (which technically is not a problem, just an implementation of the sip
> protocol that you don't typ
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
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On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
wrote:
> On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese wrote:
>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>> and Google Voice?
>>
> I wrote one last week:
> http://blog.polybeacon.com/2010/10
On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese wrote:
> On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
> wrote:
>> On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese wrote:
>>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>>> and Google Voice
On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina wrote:
> On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
>> Evening,
>>
>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>> and Google Voice?
>>
>> Thanks
>
> For
> Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc.,
> for outbound calls, it acts basically like a fancy click-to-call application.
>
> So...
>
> You need Asterisk to "login" into GV, and "initiate" the call. GV will dial
> the number you tell it to, then connect it to on
> I keep the AGI in Git as a version control system. But, you can view the AGI
> source here:
>
> http://messinet.com/trac/browser/gv/gv.agi
>
> And at the very bottom of that page is a link to download it as an individual
> file here:
>
> http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e
> Had the same issue, but have not had a chance to find a good solution.
> You could change your status to DND. I tried invisible put seems not
> to be supported.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
> Blog:
The host I am working with has two accounts from the same DID
provider. Incoming calls work correctly and dial the appropriate
extensions. This also allows incoming calls to be billed appropriately to the
individual DID accounts.
Outgoing calls from either extension default to the first DID, i.e.
> The outgoing caller-id is probably just the extension number, so the
> provider is setting it to a default (usually the main billing number). You
> can set what Asterisk sends as the outbound Caller-ID in the outbound
> context before the Dial statement. Make sure your provider will honor what
> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf You can
> use the CUT func
>On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell
>wrote:
> You can also just use an agi script to look up their current caller-id in a
> database, and set it to the correct caller-id needed.
>
> exten => _NXXNXX,1,AGI(getcid.pl,${CALLERID(NUM)},1)
> exten => _NXXNXX,n,Dial(SIP/+1${ext.
On Sun, Dec 19, 2010 at 2:40 PM, Joshua Colp wrote:
> I'm surprised nobody has suggested using the setvar functionality. It's
> extremely
> useful for stuff like this and would allow you to keep all CallerID
> information
> with the actual configuration of the device.
>
> Using a configuration e
>> First, when using multiple accounts from the same DID provider, is it
>> ideal to use IP based routing using one context as I currently am or
>> have a separate contexts for each account in the sip.conf?
>
> That's really the only way to do it presently.
So I should have multiple incoming and o
>> So I should have multiple incoming and outgoing contexts? Vitelity
>> will allow me to use IP routing or user/pass auth, the latter would
>> allow me to specify the outgoing context, this would also guarantee
>> the correct account is billed and not alone rely on caller-ID.
>
> Let me clarify fu
On Sun, Dec 19, 2010 at 4:36 AM, Jeroen Eeuwes wrote:
> Hi Stephen,
>
>> Thanks for the heads up, I have been setting the caller-ID but the
>> trouble I'm running into is specifying the which number to call out
>> as. How can an extension specify a different number? See below for my
>> current ext
> I believe I have made a little headway. I have two outgoing DID
> contexts and have changed the GotoIf statement to the extension name.
> User One acts as expected and User two now displays unknown when
> calling so I believe it is trying to to goto 20 but it's not quite
> making it. Any tips? Th
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
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On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese wrote:
> Is there a way to include:
>
> _NXXNXX
> _NXX
> _011.
> _911
>
> into my current plan:
>
Sorry, here's the rest.
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX
On Wed, Dec 22, 2010 at 2:01 AM, Jeroen Eeuwes wrote:
> Hi Stephen,
>
>> _NXXNXX
>> _NXX
>> _011.
>> _911
>
> Of course it can, but it depends on what you want to do when those
> numbers are called...
>
> I didn't know about the setvar in the sip.conf and actually I think it
> is a much "c
On Wed, Dec 22, 2010 at 12:59 PM, Warren Selby wrote:
> On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese wrote:
>>
>> On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese wrote:
>> > Is there a way to include:
>> >
>> > _NXXNXX
>> > _NXX
> To answer your first question - ${MACRO_EXTEN} is a macro-specific
> variable. It's the ${EXTEN} that called the macro, since using ${EXTEN}
> inside a Macro would just give you a value of "s".
>
> As for your second question, that's pretty easy to do. If every outbound
> call needs to be forma
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