Posted: Sat May 16, 2009 10:53 am Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully? Attached are some confs.
Thanks a lot for your assistance.
Kind Regards,
Wilson
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Sat May 16, 2009 11:30 am Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On 16 May 2009, at 12:46, Timothy Smith wrote:
Quote:
<blah>
Has anyone had the above set up working successfully? Attached are
some confs.
Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was 'insecure=very' but that
might be deprecated by now..
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Sat May 16, 2009 11:52 am Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thanks Steve for this tip.
I have insecure=very is not yet deprecated. I have added it but still no good.
I personally think the problem could be with the codecs. Any ideas?
I have attached some debug info.
Regards,
Tim
On Sat, May 16, 2009 at 3:25 PM, Steve Howes <steve@geekinter.net> wrote:
Quote:
On 16 May 2009, at 12:46, Timothy Smith wrote:
> <blah>
>
> Has anyone had the above set up working successfully? Attached are
> some confs.
>
> Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was 'insecure=very' but that
might be deprecated by now..
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Sat May 16, 2009 12:08 pm Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith@gmail.com> wrote:
Quote:
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully?
Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?
If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.
At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.
you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit
A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Sat May 16, 2009 1:30 pm Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
David,
Thanks a lot for your input. I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten => _X.,1,Read(NUM,beep,4,2,3)
exten => _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
cs-intranet*CLI>
== Using SIP RTP CoS mark 5
-- Executing [730732888@default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack
-- Accepting a maximum of 4 digits.
== Using SIP RTP CoS mark 5
-- Executing [730732888@default:1]
Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack
-- Accepting a maximum of 4 digits.
cs-intranet*CLI>
Thanks alot for your assistance.
On Sat, May 16, 2009 at 4:02 PM, David Backeberg <dbackeberg@gmail.com> wrote:
Quote:
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith@gmail.com> wrote:
> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> Has anyone had the above set up working successfully?
Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?
If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.
At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.
you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit
A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Sat May 16, 2009 2:57 pm Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Steve Howes schrieb:
Quote:
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was 'insecure=very' but that
might be deprecated by now..
insecure=very should now be written as insecure=port,invite
Posted: Sun May 17, 2009 11:48 am Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith <timotsmith@gmail.com> wrote:
Quote:
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten => _X.,1,Read(NUM,beep,4,2,3)
exten => _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Mon May 18, 2009 8:07 am Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding reverse dial-peer and the DTMF conf that you said.
However, I have checked my side and all seems to be ok. I've also
tried changing the dtmfmode to sip-notify on the gateway (and info in
sip.conf) but no luck!
Please look at the attached and give me some pointers.
Thanks,
Tim
On Sun, May 17, 2009 at 3:44 PM, David Backeberg <dbackeberg@gmail.com> wrote:
Quote:
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith <timotsmith@gmail.com> wrote:
> I have finally managed to get voice working. I both parties can hear
> each other. The problem was nating. Our network is fairly big and
> these machines are atleast 2 switches from each other. I just enabled
> it (nat=route or nat=yes) and it worked.
>
> It's not yet done however. When I redirect a call to any Asterisk
> application, it just hangs up! I have read some history and archives,
> but none of the solutions has worked for me. e.g ip inspect udp
> idle-time 900. My router (or IOS) doesn't have thet command.
>
> Could you please assist point to what could be causing this and how to
> solve it? Below are some logs and attached is the router log.
>
> ; This is the extension conf. Enter the extension you want to reach
> now (something like auto attendant).
> exten => _X.,1,Read(NUM,beep,4,2,3)
> exten => _X.,n,Dial(SIP/${NUM})
>
> ; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
_______________________________________________
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Posted: Tue May 26, 2009 9:47 pm Post subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number through a cisco router with a channel bank.. Audio worked well.. i setup a dial plan in asterisk to Dial(${EXTEN}@ciscoip) and authorise the cisco router's ip on the asterisk server and treat the calls comming from it like any other SIP calls inside the server..
-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully? Attached are some confs.
Thanks a lot for your assistance.
Kind Regards,
Wilson
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