Re: [asterisk-users] Simplifying dial-plan
To answer your first question - ${MACRO_EXTEN} is a macro-specific
variable. It's the ${EXTEN} that called the macro, since using ${EXTEN}
inside a Macro would just give you a value of s.
As for your second question, that's pretty easy to do. If every outbound
call needs to be formatted in the format 1NXXNXX, you would do this
(again, untested, but should be good along with the macro I gave you
earlier):
[globals]
DEFAULT_AREA_CODE=555 ; swap with your default area code
[outbound-context]
exten = _1NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _NXXNXX,1,Goto(outbound-context,1${EXTEN},1)
exten = _NXX,1,Goto(outbound-context,1${DEFAULT_AREA_CODE}${EXTEN},1)
exten = _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = 911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
Thanks again Warren, that works quite well!
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Re: [asterisk-users] Simplifying dial-plan
On Wed, Dec 22, 2010 at 2:01 AM, Jeroen Eeuwes jeroeneeu...@gmail.com 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 cleaner solution. Since you are already using it I would
suggest to not only use it for CallerID but also for the
@vitel-outbound like this:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@${outbound})
exten = _1NXXNXX,n,Goto(h,1)
Of course you'll need to set setvar=Outbound=vitel-outbound or
setvar=Outbound=vitel-outbound2 in sip.conf.
What do you want to do with the other numbers? If you want to do the
same as with _1NXXNXX you can just add things like this in your
extensions.conf:
exten = _NXXNXX,1,Goto(_1NXXNXX,1)
exten = 911,1,Goto(_1NXXNXX,1)
Or you can do different things if you want that like this:
exten = _NXX,1,Set(CALLERID(all)=No one cares 0)
exten = _NXX,n,Dial(SIP/${ext...@abcdefgh)
exten = _NXX,n,Goto(h,1)
Best regards,
Jeroen Eeuwes
Jeroen, I'm trying to avoid rewriting the outgoing block for the
patterns mentioned above. I've placed a pseudo dial-plan below. The
plan needs to dial the 1 and/or also the area code depending on the
pattern they enter. Any tips, thanks.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _NXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _NXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _NXXNXX,n,Goto(h,1)
exten = _NXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
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Re: [asterisk-users] Simplifying dial-plan
On Wed, Dec 22, 2010 at 12:59 PM, Warren Selby wcse...@selbytech.com wrote:
On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese rsre...@gmail.com wrote:
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com 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=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Why not make a Macro (or GoSub) to handle this block of code, and then your
outbound dial lines are just one line calling the Macro? Saves a lot of
repeating blocks of code. Something like this (not tested):
[macro-OutboundDial]
; ${ARG1} = CHANNEL
; ${ARG2} = EXTERNAL_CALLERID
exten = s,1,Set(Outgoing=${CUT(${ARG1},/,2)})
exten = s,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = s,n,GotoIf($[${Outgoing} = 201]?outbound2:outbound1)
exten = s,n(outbound1),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound)
exten = s,n,Goto(h,1)
exten = s,n(outbound2),Set(CALLERID(all)=${ARG2})
exten = s,n,Dial(SIP/${macro_ext...@vitel-outbound2)
exten = s,n,Goto(h,1)
[outbound-context]
exten = _NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _NXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
exten = _911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
Thanks Warren, that's what I'm looking to do.
First question is where did ${MACRO_EXTEN} come from, I assumed
${EXTEN} is a built in variable?
Secondly, where would the 1 and/or area-code need to be placed? Could
an additional argument be used to specify the prefix, i.e. a third
variable be specified in the outbond-context to implement the
OutboundDial macro, or is the MACRO_EXTEN suppose to be an
implementation of this?
exten = s,n,Dial(SIP/{$arg3}${macro_ext...@vitel-outbound2)
As Jeroen mentioned previously a goto may be used, would this help,
seems similar to what I am trying to accomplish.
exten = _NXXNXX,1,Goto(1${EXTEN},1)
exten = _NXX,1,Goto(1555${EXTEN},1)
Thanks,
Stephen
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[asterisk-users] Simplifying dial-plan
Is there a way to include: _NXXNXX _NXX _011. _911 into my current plan: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simplifying dial-plan
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Thanks
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Re: [asterisk-users] Specifying DID for outbound calls
You can check the channel-name to see which extension is making the
call and set the CallerID accordingly. The channel-name will be
something like SIP/201-abc23ef34 or SIP/User1-def34abc51. The 201
or User1 part depends on how you put the username in sip.conf You can
use the CUT function to get the calling extension and then jump to the
correct CallerID. I've used something like this:
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = User2]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten = _1NXXNXX,n,Set(CALLERID(name)=User1)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten = _1NXXNXX,n,Set(CALLERID(name)=User2)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
But in my case I had two different domains. E.g.
Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
instead of setting the CallerID.
Not that the Cut doesn't work correctly if you use a minus-sign in the
username.
Best regards,
Jeroen Eeuwes
Thanks Jeroen, though it is still not firing correct, I have provided
a little more information.
Here are the channel-names:
SIP/201-000a
SIP/101-0012
Here is the extension information from the sip.conf:
[101]
type=friend
username=101
secret=
mailbox=101
callerid=User One 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[201]
type=friend
username=201
secret=
mailbox=201
callerid=User Two 201
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
Here is the updated outgoing context that you provided with a few updates.
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = User Two]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten = _1NXXNXX,n,Set(CALLERID(name)=User One)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten = _1NXXNXX,n,Set(CALLERID(name)=User Two)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
Based on the information above, what should be altered to correctly
associated the number with the relevant extension?
Thanks
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Re: [asterisk-users] Specifying DID for outbound calls
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 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. Thanks for being responsive, I do not work with Asterisk much, actually I do not touch it unless I need to add more functionality outside of regular patching so my fu is not strong in this area ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
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 further. For calls FROM vitelity you are pretty much limited to a single context in sip.conf doing IP based matching. Most equipment will not authenticate to you, and chan_sip currently has no additional method for separating the accounts into separate contexts. For calls TO vitelity you should probably have separate contexts. Thanks for being responsive, I do not work with Asterisk much, actually I do not touch it unless I need to add more functionality outside of regular patching so my fu is not strong in this area ;-) -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org Great, I'll get it changed and see if it helps, thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 4:36 AM, Jeroen Eeuwes jeroeneeu...@gmail.com 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 extension.conf, thanks.
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 function to get the calling extension and then jump to the
correct CallerID. I've used something like this:
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = User2]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten = _1NXXNXX,n,Set(CALLERID(name)=User1)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten = _1NXXNXX,n,Set(CALLERID(name)=User2)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
But in my case I had two different domains. E.g.
Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
instead of setting the CallerID.
Not that the Cut doesn't work correctly if you use a minus-sign in the
username.
Best regards,
Jeroen Eeuwes
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=User One 3012323434)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=User Two 3013232322)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
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Re: [asterisk-users] Specifying DID for outbound calls
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=User One 3012323434)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=User Two 3013232322)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
Disregard, I had num instead of all for the CALLERID statement.
Thanks for all of the help!
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[asterisk-users] Specifying DID for outbound calls
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. calls from either extension have the same callerID. How can an extension specify separate outgoing contexts so the correct number is associated with it, also allowing the SIP provider to recognize the difference for billing purposes, or is there a better way? In short I'm looking to associate an outgoing call from an extension with a specific number. Here's the sip.conf for both accounts as they are using IP routing, I'm assuming I do not have to perform auth based to separate the two accounts for outgoing calls: [vitel-inbound] type=friend dtmfmode=auto host=inbound18.vitelity.net context=inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=outbound insecure=very allow=all Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
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 set, as many filter what you can send to only the DIDs they provide for
you.
Take a look here for more information on setting the caller-id in the
dialplan:
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
-Jonathan
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 extension.conf, thanks.
[default]
exten = 201,1,Dial(SIP/201@,30)
exten = 201,n,Voicemail(2...@default)
exten = 201,n,Hangup
exten = 202,1,Dial(SIP/202,30)
exten = 202,n,Voicemail(2...@default)
exten = 202,n,Hangup
include = inbound
include = outgoing
[inbound]
exten = 3012323434,1,Goto(default,201,1)
exten = 3013232322,1,Goto(default,202,1)
[outgoing]
exten = _1NXXNXX,1,Set(CALLERID(num)=3012323434)
exten = _1NXXNXX,n,Set(CALLERID(name)=User1)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
[outgoing2]
exten = _1NXXNXX,1,Set(CALLERID(num)=3013232322)
exten = _1NXXNXX,n,Set(CALLERID(name)=User2)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
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/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi This is not the same thing as what is in the Changelog. I am using Asterisk 1.6 with this AGI. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Thanks Anthony, Interestingly enough outbound dialing started working. Had no clue until someone called and told me my Google Chat status was updated. Is there a way to prevent Google Chat from staying logged in but still be able to dial outbound? People think I'm logged in persistently and send me messages that I miss. Even if I set a status message in asterisk most users are not going to understand... -Stephen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
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 one of your GV numbers. In my case, the AGI is what connects to GV and initiates the call. GV, then dials the number I told it to dial, then connects it with my ipKall number (which I have as one of my GV numbers). In Asterisk, the outbound call runs the AGI and places the channel in the DB, then waits for an incoming call via my inbound ipKall trunk. Once the ipKall comes into Asterisk, the Bridge command is used to bridge the original (with the matching DB entry) call-- the call that is coming in from GV through ipKall. I suppose you don't need that AGI and could probably do this using Curl in the dialplan. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E That makes sense but I do not see where the new feature is in Asterisk 1.8 which include Google Voice support per http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt: 290973 |dvossel|Make outbound Google Voice calls. | | It seems that the GV has been a feature for sometime with previous versions? I'm just trying to keep the process as simple as possible and seeing three different methods is a little confusing: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ (no script referenced) http://www.davidvossel.com/?p=28 (python script and listed in the change log above) http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script) Is your .agi and .git the same script? I do not have a git client on this host to see for myself. Thanks, Stephen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina amess...@messinet.com 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 Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
; inbound ipKall trunk (to which Google Voice makes the connection)
[ipkall]
exten = ipKall-number,1,GotoIf($[${DB_EXISTS(gv/channel)} = 1]?gv)
same = n,Goto(default,s,1)
same = n(gv),Bridge(${DB_DELETE(gv/channel)})
same = n,AGI(gv/gv.agi,hangup)
same = n,Hangup()
; outbound Google Voice initiation
[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
Does the AGI have to be used? In this example
http://www.davidvossel.com/?p=28 I see mention of a script, but not in
this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
I believe I missing the connection in how the whole process actually
works therefore making troubleshooting a little difficult. I was
hoping with the release of 1.6.0 there wouldn't be a lot of bandage
work to get it to play nicely with Google Voice.
--
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[asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com 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/2010/10/17/asterisk-1-8-and-google-voice/ Also: http://www.davidvossel.com/?p=28 Paul, It seems you were using a beta/SVN release for your example. Do the following two packages need to be installed if using the stable 1.6.0 release before building from source? I ask as I am unable to dial out. $ apt-get install libikesemel-dev $ apt-get install libssl-dev Secondly, do you know if the username/password are sent in clear text to the Google? Thanks, Stephen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese rsre...@gmail.com wrote: On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com 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/2010/10/17/asterisk-1-8-and-google-voice/ Also: http://www.davidvossel.com/?p=28 Paul, It seems you were using a beta/SVN release for your example. Do the following two packages need to be installed if using the stable 1.6.0 release before building from source? I ask as I am unable to dial out. $ apt-get install libikesemel-dev $ apt-get install libssl-dev Secondly, do you know if the username/password are sent in clear text to the Google? I installed the two packages previously mentioned but still lack outbound dialing. I enabled debugging and am getting the following messages. I double checked the password and even changed it to one without special characters but still the same results. JABBER: gmail INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1693 aji_act_hook: JABBER: encryption failure. possible bad password. JABBER: gmail INCOMING: /stream:stream [Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1576 aji_act_hook: aji_act_hook was called with out a packet [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:1391 aji_recv: Parsing failure: Hook returned an error. [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2742 aji_recv_loop: JABBER: Got hook event. [Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2753 aji_recv_loop: JABBER: socket read error -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wcse...@selbytech.com 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 typically see). As to your issue, are you still seeing the same error messages in the ssh logs? I haven't ever had to use the register with proxy settings in my configs, but I've only worked with the 79x1 series phones, not the x2. I've actually got a post up on my blog addressing setting up a 7941 in a situation similar to yours: http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/ In that post is a sanitized version of my conf file that I use on my own deskphone, if you'd like to download it and try it out with your setup. My config is very similar though my only question is you have registerWithProxy set to true though nothing defined. Was this a sanitation mistake? sipProxies backupProxy/ backupProxyPort/ emergencyProxy/ emergencyProxyPort/ outboundProxy/ outboundProxyPort/ registerWithProxytrue/registerWithProxy /sipProxies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't configure Cisco 7942 avec factory reset
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+SIP http://www.greenwireit.com/blog/2009/09/cisco-7961-and-7941-sip-configuration-sepmac-cnf-xml/ On Thu, Oct 22, 2009 at 10:48 AM, Olivier oza-4...@myamail.com wrote: Hi, (I think) I followed instructions here (http://www.voip-info.org/wiki/view/Firmware+issues+on+7940+-+7960 section Notes added Nov 2005, revised May 2006: at the bottom of the page) to factory reset a Cisco 7942 I wanted to configure to SIP firmware. When booting, I can see this requesting and obtaining file term42.default.loads from TFTP server. Then it would send a request (recognized as CDP request by Wireshark) a couple of times, then loop again asking for term42.default.loads file. My question is : Is it possible to upload a SIP firmware with a factory reset 7942, without any Call Manager ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Tue, Nov 10, 2009 at 10:13 AM, Warren Selby wcse...@selbytech.com 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 asterisk send the way the phone will accept. Thanks, --Warren Selby Thanks, as a test I changed both a 7960 and 7942 both to nat=no the latter being the one I'm having trouble registering. The 7960 then was unable to register so I changed it back to nat=yes. When I changed the 7942 to nat=no and disabled registerWithProxy I can get a dial tone but can't dial out due to the following: SIP/2.0 407 Proxy Authentication Required I tried re-enabling the proxy but then I get nothing as before. From the sip.conf I would assume that registerWithProxy would be the same as the realm and the proxy statement for the phone be the same as the domain? realm=ns1.domain.net domain=domain.net Thanks again for any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby wcse...@selbytech.com 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 log files which will show you your error. Thanks for the insight. After you mentioned that the syntax of the XML file may be wrong I looked around and found a more complete configuration I could find since mine was a copy and paste special. Using the new configuration the phone comes up but is unable register I *think* it may be an issue with NAT. When the phone fires up for the first time it tries to register for a while and the log didn't help much so I took a peak at the asterisk logging. It seems like packets are not getting back to the phone. I've enabled NAT in the configuration similar to how the other phones are configured but no dice. Note that the Asterisk device is not NATed but the phones are behind a NAT device. I get multiple of the following message in the phone: ERR 16:40:16.273722 JVM: %REG send failure: REGISTER On the asterisk server I keep getting NAT retries: Retransmitting #4 (NAT) to 71.226.175.137:1026: OPTIONS sip:1...@ip of NAT device:1027;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP ASTERISK IP:5060;branch=z9hG4bK53121c03;rport From: asterisk sip:aster...@209.251.157.91;tag=as5b0b32f5 To: sip:1...@ip of NAT:1027;user=phone;transport=udp Contact: sip:aster...@209.251.157.91 Call-ID: 090e1e583f29f9f000dd30ff5719f...@209.251.157.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Nov 2009 02:26:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Below is the full XML config for the phone: device xsi:type=axl:XIPPhone ctiid=9044468655 deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool dateTimeSetting dateTemplateM/D/Ya/dateTemplate timeZoneEastern Standard/Daylight Time/timeZone ntps ntp name192.43.244.18/name ntpModedirectedbroadcast/ntpMode /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameAsterisk IP/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies backupProxy/backupProxy backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxyAsterisk IP/outboundProxy outboundProxyPort5060/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures sipStack sipInviteRetx6/sipInviteRetx sipRetx10/sipRetx timerInviteExpires180/timerInviteExpires timerRegisterExpires3600/timerRegisterExpires timerRegisterDelta5/timerRegisterDelta timerKeepAliveExpires120/timerKeepAliveExpires timerSubscribeExpires120/timerSubscribeExpires timerSubscribeDelta5/timerSubscribeDelta timerT1500/timerT1 timerT24000/timerT2 maxRedirects70/maxRedirects remotePartyIDfalse/remotePartyID userInfoNone/userInfo /sipStack autoAnswerTimer1/autoAnswerTimer autoAnswerAltBehaviorfalse/autoAnswerAltBehavior autoAnswerOverridetrue/autoAnswerOverride transferOnhookEnabledfalse/transferOnhookEnabled enableVadfalse/enableVad preferredCodecg711ulaw/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandavt/dtmfOutofBand alwaysUsePrimeLinefalse/alwaysUsePrimeLine alwaysUsePrimeLineVoiceMailfalse/alwaysUsePrimeLineVoiceMail kpml3/kpml natEnabledtrue/natEnabled natAddressIP outside of NAT
Re: [asterisk-users] Trouble registering Cisco 7942
On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby wcse...@selbytech.com 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 config is giving it grief. Once you know that, you'll know what's causing the Unprovisioned message. I set the username and password but am unable to log into the phone. I provided an updated config below. I am prompted for the username and password though. Secondly should I be using IP or hostnames for the proxy and processNodeName or does it not matter? Thanks device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameSIPSERVER/processNodeName /callManager /member /members /callManagerGroup /devicePool sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures natEnabledtrue/natEnabled natAddress172.16.2.1/natAddress phoneLabel102/phoneLabel sipLines line button=1 featureID9/featureID featureLabel102/featureLabel contact102/contact proxySIPSERVER/proxy port5060/port name102/name displayNameAtlas/displayName authName102/authName authPasswordPASS/authPassword sharedLinefalse/sharedLine /line /sipLines /device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
It's definitely just for fun, I wouldn't think to try to implement such as setup for a client unless I were really comfortable with the setup! On Fri, Oct 24, 2008 at 8:36 AM, David Gibbons [EMAIL PROTECTED] wrote: Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that'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-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) 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 hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a 400 Bad request back to Asterisk. Both of these phones are on my local lan but the Asterisk server is at a colo facility on the internet outside of the local lan. The local lan does use NAT/PAT. I see an error Warning: 399 Bad Request - 'Malformed/Missing FROM: field'. Is this a problem? Thanks --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 274 Content-Type: application/sdp v=0 o=102 157742 157742 IN IP4 172.16.2.18 s=Cisco 7912 SIP Call c=IN IP4 68.156.63.118 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 - --- (14 headers 12 lines) --- Sending to 68.156.63.118 : 1083 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (NAT) to 68.156.63.118:1082 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net, nonce=7c2e1ba9 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '102' --- SIP read from 64.2.142.116:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 - --- (10 headers 0 lines) --- --- SIP read from 64.2.142.116:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED];tag=as7a2f92a1 Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec Content-Length: 0 - --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name inbound18.vitelity.net REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 64.2.142.116:5060: REGISTER sip:inbound18.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport From: sip:[EMAIL PROTECTED];tag=as751cb0af To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3065 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=rsreese, realm=asterisk, algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec, response=b765dbdebba8af18b19707efe651d65d Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from 68.156.63.118:1082 --- ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Content-Length: 0 - --- (9 headers 0 lines) --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest
Re: [asterisk-users] adding a second extension
I am able to now call the second extension when setup like this so I
believe I'll leave it alone for a while. Basically added the extension
102 to the main incoming line:
exten = 101,1,Dial(SIP/101SIP/102SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,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 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.
Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
sending a 400 Bad request back to Asterisk.
Both of these phones are on my local lan but the Asterisk server is at
a colo facility on the internet outside of the local lan. The local
lan does use NAT/PAT. I see an error Warning: 399 Bad Request -
'Malformed/Missing FROM: field'. Is this a problem?
Thanks
---
ns1*CLI
--- SIP read from 68.156.63.118:1082 ---
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157742 157742 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (14 headers 12 lines) ---
Sending to 68.156.63.118 : 1083 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
--- Reliably Transmitting (NAT) to 68.156.63.118:1082 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net,
nonce=7c2e1ba9
Content-Length: 0
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000
ms (Method: INVITE)
Found user '102'
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
-
--- (10 headers 0 lines) ---
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED];tag=as7a2f92a1
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec
Content-Length: 0
-
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name
inbound18.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.116:5060:
REGISTER sip:inbound18.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
From: sip:[EMAIL PROTECTED];tag=as751cb0af
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=rsreese, realm=asterisk,
algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec,
response=b765dbdebba8af18b19707efe651d65d
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
---
--- SIP read from 68.156.63.118:1082 ---
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP
[asterisk-users] Asterisk and Cisco Call Manager Express (CME)
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... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
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 hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
I also tried downgrading to version 1.4-current but that didn't help. Any other ideas? I'm at a loss. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
I am now using a Cisco phone for the second extension (102). I am able to contact 102 from 101 but not the other way around. The error seems less severe now: == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-0825b118, SIP/101/20) in new stack == Using SIP RTP CoS mark 5 -- Called 101/20 -- SIP/101-08221a78 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-0825b118, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-0825b118' So maybe it's just a config issue now? [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [default] exten = 101,1,Dial(SIP/101/20) exten = 101,n,Hangup exten = 101,n,Voicemail([EMAIL PROTECTED]) exten = 102,1,Dial(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 = _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)=9045622082) exten = _NXX,n,Set(CALLERID(name)=Stephen Reese) exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED]) exten = _NXXNXX,1,Set(CALLERID(num)=9045622082) exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese) exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]) exten = _011.,1,Set(CALLERID(num)=9045622082) exten = _011.,n,Set(CALLERID(name)=Stephen Reese) exten = _011.,n,Dial(SIP/[EMAIL PROTECTED]) exten = _911,1,Set(CALLERID(num)=9045622082) exten = _911,n,Set(CALLERID(name)=Stephen Reese) exten = _911,n,Dial(SIP/[EMAIL PROTECTED]) On Mon, Oct 20, 2008 at 11:06 AM, Stephen Reese [EMAIL PROTECTED] wrote: 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() instead of: exten = 102,1,Dial... But this is a very strange error... Check if there is no other definition of default having 102 on it because Asterisk is going to merge the extensions. I get the following when trying to dial 1002 from 101. I've attached my extensions.conf file in-case there is something else that is conflicting as you mentioned. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-082aca90, SIP/102/20) in new stack == Using SIP RTP CoS mark 5 -- Called 102/20 [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-082aca90, ) in new stack == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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 trunk called 101. 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 RTP CoS mark 5 -- Called 102 -- SIP/102-08221a78 is ringing -- SIP/102-08221a78 answered SIP/101-08220318 -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78 == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318' Failed call from 102 to 101 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08221a78, SIP/101,20) in new stack == Using SIP RTP CoS mark 5 -- Called 101 -- Got SIP response 400 Bad Request back from 68.156.63.118 -- SIP/101-0821e110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08221a78, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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 RTP CoS mark 5 -- Called 102 -- SIP/102-08221a78 is ringing -- SIP/102-08221a78 answered SIP/101-08220318 -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78 == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318' Failed call from 102 to 101 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08221a78, SIP/101,20) in new stack == Using SIP RTP CoS mark 5 -- Called 101 -- Got SIP response 400 Bad Request back from 68.156.63.118 -- SIP/101-0821e110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08221a78, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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('SIP/
I'm surprised nobody else noticed this.
I looked through my extensions.conf and sip.conf which are posted in
this thread I believe and didn't turn up anything significant? Would
NAT pose a problem for more then one phone behind a NAT router?
___
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Re: [asterisk-users] adding a second extension
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 Dial SIP/101 ns1*CLI originate SIP/102 application Dial SIP/102 ns1*CLI == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/102-0824a330 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-0824a330 requested special control 16, passing it to SIP/102-082256c0 -- Started music on hold, class 'default', on SIP/102-082256c0 -- SIP/102-082256c0 answered SIP/102-0824a330 -- Packet2Packet bridging SIP/102-0824a330 and SIP/102-082256c0 -- Stopped music on hold on SIP/102-082256c0 ns1*CLI originate SIP/101 application Dial SIP/102 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/101-08249e28 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-082256c0 answered SIP/101-08249e28 -- Packet2Packet bridging SIP/101-08249e28 and SIP/102-082256c0 ns1*CLI originate SIP/102 application Dial SIP/101 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/101) on SIP/102-08254038 == Using SIP RTP CoS mark 5 -- Called 101 -- SIP/101-08252a40 is ringing -- SIP/101-08252a40 answered SIP/102-08254038 -- Packet2Packet bridging SIP/102-08254038 and SIP/101-08252a40 So I the two extensions are able to call each other with the later two sets of commands so there is hope :-). Would my NAT have anything to do with it since I'm specifying the proxy host that is outside of my firewall? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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() instead of: exten = 102,1,Dial... But this is a very strange error... Check if there is no other definition of default having 102 on it because Asterisk is going to merge the extensions. I get the following when trying to dial 1002 from 101. I've attached my extensions.conf file in-case there is something else that is conflicting as you mentioned. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-082aca90, SIP/102/20) in new stack == Using SIP RTP CoS mark 5 -- Called 102/20 [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-082aca90, ) in new stack == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90' extensions.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latency woes, qos the fix?
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 traffic to is just that far away in latency terms. Interesting. Just to clarify, the computer I'm pinging from is on the same switch as the phone so I don't see how there could be so much variance since the remote Asterisk server is on a very fast pipe. I've seen several people post that they have well under 100ms response. Is the time that the Asterisk displays just a ping to the device or are there more calculations? Any ideas besides TOS since that will not help much as you mentioned? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latency woes, qos the fix?
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. TCP traffic simply retransmits, but voice (on asterisk) is RTP/UDP and the packet gets dropped. Even if it were TCP there is no time for a retransmit to be detected and resent. Using ehternet to detect the collision it does get resent, but there comes your jitter - which has much worse effects than simply latency. As far as measuring latency, doing a sip show peer andlooking at the qualify times is a GUIDELINE. It is my no means a correct indication, the real time can be much lower. I have noticed various ATA on the same networks as Polycom phones wil have sub 20ms times and the Polycoms will be 50ms. Yet all is as it should be and working great. Generally QOS will help with packet loss and jitter. Hope this helps. You were both right I was just double checking. I fired up a soft phone on a desktop that has relatively low ping rates and experienced similar response times ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.29 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 102/10268.156.63.118D N 56558OK (145 ms) 101/10168.156.63.118D N 1038 OK (135 ms) Thank you both for your insight. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
;include = demo
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,Dial(SIP/102,20)
exten = 102,n,Hangup
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,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 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 = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
This is a call from extension 101 to 102 that fails with a busy signal.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08266f60,
'SIP/102',20) in new stack
[Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08266f60, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60'
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Re: [asterisk-users] adding a second extension
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 just fine using 101. It's the extension to extension calling from 101 to 102 and vice-versa that is not working. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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 from 101 to 102 and 102 to 101.
Having the sip.conf sould help on getting whats going on.
Here are the relevant parts of the sip.conf
[general]
register = rsreese:[EMAIL PROTECTED]:5060/rsreese
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
;match_auth_username=yes; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
;allowtransfer=no ; Disable all transfers (unless
enabled in peers or users)
; Default is enabled
realm=ns1.neocipher.net ; Realm for digest authentication
; defaults to asterisk. If you set a
system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard
port for unencrypted UDP
; and TCP sessions is 5060)
; bindport is the local UDP port that
Asterisk will listen on
bindaddr=0.0.0.0; IP address to bind UDP listen socket
to (0.0.0.0 binds to all)
; You can specify port here too, like
123.123.123.123:5080
domain=neocipher.net
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=pass
mailbox=101
callerid=\Stephen\ 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=pass
mailbox=102
callerid=\Stephen\ 102
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=test
allow=all
;insecure=very
insecure = invite
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=test
allow=all
canreinvite=no
Here is the sip debug error:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08266f60,
'SIP/102',20) in new stack
[Oct 19 16:21:21] WARNING[26690]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 16:21:21] WARNING[26690]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08266f60, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/102-08266f60'
Scheduling destruction of SIP dialog
'NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.' in 32000 ms (Method:
INVITE)
ns1*CLI
--- Reliably Transmitting (NAT) to 68.156.63.118:56558 ---
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
68.156.63.118:56558;branch=z9hG4bK-d8754z-e0a3d830adb35401-1---d8754z-;received=68.156.63.118;rport=56558
From: sip:[EMAIL PROTECTED];tag=7d39014c
To: 102sip:[EMAIL PROTECTED];tag=as4f32f2a7
Call-ID: NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
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Re: [asterisk-users] adding a second extension
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 look correct? I tried reinstalling and I still am unable to communicate between the two extensions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
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 what happens. If you got a CVS commit, commit again or try installing a release. http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for download) Regards, Juan I grabbed the latest tarball and installed it. The extension rings through to 101 and then when I answer it tries to ring through to 102 but seems to fail. ns1*CLI originate SIP/101 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08245390, 'SIP/102',20) in new stack [Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08245390, ) in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390' The extension rings through to 102 and when I answer the line it begins to ring line 101. ns1*CLI originate SIP/102 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08249e28, SIP/101SIP/[EMAIL PROTECTED],30) in new stack == Using SIP RTP CoS mark 5 -- Called 101 == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- SIP/101-08244e88 is ringing -- SIP/vitel-outbound-0825d1e0 is making progress passing it to SIP/102-08249e28 -- SIP/vitel-outbound-0825d1e0 is ringing -- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28 -- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0 == Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28' I'm at a loss. Thanks for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
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 device, allowing calls to come in until it expires (default on many devices is 60 seconds). You may also receive inbound calls when the phone reregisters regularly. Try 'qualify=yes' in your phones section in sip.conf to send keepalives (option packets in this case) every two seconds to the phone to keep it from going idle. You can see the state of the phone from the console with a 'sip show peers', if unreachable, your NAT device has killed the NAT forward. Should look like one of these: xxx/xxx x.x.x.x D N 5060 OK (46 ms) xxx/xxx x.x.x.x D N 5060 UNREACHABLE As another troubleshooting step, you can telnet to the phone and have it reregister with Asterisk manually (register line 1 1) to see if that brings it back to life. If qualify doesn't do it, see if you can increase UDP timeouts in your firewall/NAT device. I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Very cool, I believe that did the trick. Thank you for your time. On Sat, Oct 18, 2008 at 7:42 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect 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 calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0
[asterisk-users] Latency woes, qos the fix?
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 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.111D N 1038 OK (133 ms) This seems pretty high when my ping time from a host on the same network is ~30ms: Pinging 209.251.157.93 with 32 bytes of data: Reply from 209.251.157.93: bytes=32 time=30ms TTL=51 Reply from 209.251.157.93: bytes=32 time=27ms TTL=51 Reply from 209.251.157.93: bytes=32 time=36ms TTL=51 Reply from 209.251.157.93: bytes=32 time=28ms TTL=51 Any suggestions or is this normal? Should I enable qos on my Cisco 3725 router and 2950 switch? Would I also need to enable the following in the sip.conf ;tos_sip=cs3; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5; Sets 802.1p priority for RTP audio packets. ;cos_video=4; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
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 the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
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: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name: scott
#line2_authname: scott
#line2_shortname: 201
#line2_password: tiger
#line2_displayname: Larry Ellison; # Line 2 Display Name (Display
name to use for SIP messaging)
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: Stephen Reese ; Has no effect on SIP messaging
# Phone Password (Password to be used for console or telnet login)
phone_password: goaway ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
telnet_level: 2
Any ideas or help would be great, thanks.
I'm still unable to wrap my head around this problem. I can recieve a
call after I first call out from the line/phone. I didn't think it's a
NAT issue since it kind of works.
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[asterisk-users] Cisco 7960 not always receiving incoming calls
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 the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
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: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name: scott
#line2_authname: scott
#line2_shortname: 201
#line2_password: tiger
#line2_displayname: Larry Ellison; # Line 2 Display Name (Display
name to use for SIP messaging)
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: Stephen Reese ; Has no effect on SIP messaging
# Phone Password (Password to be used for console or telnet login)
phone_password: goaway ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
telnet_level: 2
Any ideas or help would be great, thanks.
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Re: [asterisk-users] Menu for call forwarding or voicemail
Any reason not to ring both at once?
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],20)
-Darren
That does work and it rings both phones. I was able to put in some
error handling to deal with the extension if the softphone isn't
connected and still allow the remote phone to ring :-).
The trick now is the timing for the remote voice mail. Since cell
phone contact times can sometimes vary in length what can I do to make
sure a users call isn't cut off when leaving a message if the ring and
voicemail message are greater then 20? Currently it just goes to a
busy tone when the 20 seconds are reached... Should I just increase
the time? I'm still concerned if someone leaves a long message that
they may get cut off?
[general]
static=yes
writeprotect=yes
[globals]
[default]
;exten = 101,1,Dial(SIP/101,20)
;exten = 101,n,Dial(SIP/[EMAIL PROTECTED])
;exten = 101,n,Voicemail([EMAIL PROTECTED])
;exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],20)
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED]|20)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED]|20)
exten = 101,n,Goto(lbl_default_0)
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,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 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 = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
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Re: [asterisk-users] Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Voicemail([EMAIL PROTECTED])
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,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 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 = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
Okay I'm going to start simple.
First I would like to forward the number to the remote number which
we'll make 904-940-9007. I've commented out the voicemail for the time
being, I'll bring that in once a menu is composed later on. So anyways
I've added a second rule to dial the second number after 20 seconds is
that the correct placement?
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Dial(SIP/[EMAIL PROTECTED])
;exten = 101,n,Voicemail([EMAIL PROTECTED])
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,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 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 = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
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Re: [asterisk-users] Menu for call forwarding or voicemail
Any reason not to ring both at once? exten = 101,1,Dial(SIP/101SIP/[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 voicemail to implement your rule then it will error out since the phone isn't connected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
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 to make outgoing calls after much deliberation. I had to
add callerid to my outgoing...
Here's the extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Voicemail([EMAIL PROTECTED])
;exten = 101,102,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM} variable
in the current context (var ${CONTEXT}).
include = outgoing
include = inbound
[outgoing]
; The following gives an Unknown Caller ID
;exten = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
; The following will display your number on a caller ID
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=9045622082)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
;exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
; e911 must be enabled. see DIDs NPANXXNXXX Action e911
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Goto(default,101,1)
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[asterisk-users] Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Voicemail([EMAIL PROTECTED])
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,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 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 = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
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Re: [asterisk-users] Vitelity Asterisk configuration help
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 and reply with any error messages. I was not adding the 1 before the number but that didn't help. I opened the console 'asterisk -r' but when attempting to call out nothing happened. Is there some type of logging level that needs to be turned up? When I call in which does still work I do get the following errors and of course voicemail doesn't work.: Oct 7 09:38:08 WARNING[6146]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '102' Oct 7 09:38:18 WARNING[6146]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Thanks again for the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
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. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
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
outbound calling include your outgoing context within the default. To
include it, at the bottom of the default context add include =
outgoing
either of these should allow outgoing calling. As for incoming, add a
Goto
as follows.
[inbound]
exten = 9045622082,1,Answer
exten = 9045622082,n,Goto(default,101,1)
That equates to goto the default context, extension 101, at the 1st
priority which is your Dial command.
Best Regards,Darren Severino
Thanks I am now able to make incoming calls but I'm still unable to call
out. Notice anything else off.
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail(102)
exten = 101,102,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;This automatically
calls the right mailbox using the ${CALLERIDNUM} variable in the current
context (var ${CONTEXT}).
include = outgoing
include = inbound
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
; e911 must be enabled. see DIDs NPANXXNXXX Action e911
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Goto(default,101,1)
Sip.conf
[general]
register = rsreese:[EMAIL PROTECTED]:5060
context=default ; Default context for incoming calls
realm=ns1.neocipher.net ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
domain=neocipher.net; Set default domain for this host
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=test81
mailbox=101
callerid=\Stephen\ 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
musicclass=default ; Sets the default music on hold class for
all SIP calls
language=en ; Default language setting for all
users/peers
[authentication]
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=key
allow=all
insecure=very
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=key
allow=all
canreinvite=no
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[asterisk-users] Vitelity Asterisk configuration help
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 incoming and outgoing calls with voicemail
support. I've searched all over but many of the full configurations
that are available are a bit complex. Any tips or recommendations to
get up and running would be great.
sip.conf
Code:
[general]
register = rsreese:[EMAIL PROTECTED]:5060
context=default ; Default context for incoming calls
realm=ns1.neocipher.net ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
domain=neocipher.net; Set default domain for this host
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=test81
mailbox=101
callerid=Stephen 101
host=dynamic
dtmfmode=rfc2833
canreinvite=no
reinvite=no
disallow=all
allow=gsm
[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=test81
mailbox=102
callerid=(Bob 101)
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
allowguest=yes
insecure=very
promiscredir=yes
musicclass=default ; Sets the default music on hold class
for all SIP calls
[authentication]
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=pass
allow=all
insecure=very
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=pass
allow=all
canreinvite=no
extensions.conf
Code:
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail(102)
exten = 102,1,Dial(SIP/102,20)
exten = 102,2,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Answer
voicemail.conf
Code:
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Context to Send voicemail from [option 5
from the advanced menu]
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
[default]
101 = 123,Stephen Rese,[EMAIL PROTECTED]
102 = 123,Bob Dole,[EMAIL PROTECTED]
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
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