[asterisk-users] bash: asterisk: command not found

[christopher kamutumwa]

Hello

Ive installed asterisk 14.2 on centos 6.8 but i am not able to start it
below is what am executing and those are the errors anything am doing wrong?

[root@localhost ~]# asterisk -r
bash: asterisk: command not found
[root@localhost ~]# asterisk -vvvc
bash: asterisk: command not found

chris
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Re: [asterisk-users] T1 -Asterisk server - Analog lines

[Doug Lytle]

On 12/06/2016 06:45 PM, Motty Cruz wrote:
Any suggestions on how to convert digital signal to analog? 


I do this with an ADIT 600 channel bank that I got off of ebaY many 
years ago.  A quick search shows one for around $98


http://www.ebay.com/itm/like/191565965269?lpid=82=ps_noapp=true

Doug

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[asterisk-users] T1 -Asterisk server - Analog lines

[Motty Cruz]

Hello All, 

The problem I'm facing is the following: two machines that require analog
signal. However, connection to the world is setup as follow: T1 connection
to Asterisk server

Any suggestions on how to convert digital signal to analog? 

Type of card on my Asterisk server is wct4xxp

 

Thanks for your support!

Motty

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Re: [asterisk-users] new inbound DID provider... no auth?

[Max Grobecker]

Hi,

That's right - you just need to define a peer with a static IP address and 
"type=peer" to assign incoming calls to a peer name and apply
the corresponding configuration (e.g. codecs).
To make your configuration less redundant you can use templates in your peer 
definition
(at least for chan_sip, I'm not sure if the same syntax applies on chan_pjsip).

Example:

---

;; All configuration made to this peer will be applied to all childs of this 
definition
[your-did-provider](!)
type=peer
allow=ulaw,alaw,g722
...

;; This peer derives all other configuration from "your-did-provider",
;; then your local changes are applied and can override the derived ones.
[your-did-provider-gw1](your-did-provider)
host=1.2.3.4

[your-did-provider-gw2](your-did-provider)
host=1.2.3.5
---

That's the shortest thing I can imagine at the moment.
At least, with this way of definition you only need to do changes on one single 
point, not for every gateway IP.





Am 30.11.2016 um 22:10 schrieb Jeff LaCoursiere:
> 
> We are trying to work with a new DID provider and I find myself confused.  
> Their standard integration is to send the call with no authentication.  I am 
> expected to whitelist all their possible gateways, and accept their calls I 
> guess with no peer definition.  I actually have it working this way; the 
> calls land in our "public" context, I guess as "guest", and I am able to 
> route them from there.  But that makes me nervous.
> 
> I would rather at least have them be associated with a defined peer, so I can 
> set the right context and any other parameters I might want associated.  It 
> is inbound only, no outbound.  I might try to set a host= in a peer 
> definition with no secret, and see if that matches it, but I would rather 
> avoid making a peer definition for every gateway they have.  Can anyone think 
> of a way to define a single peer that might show from multiple potential 
> addresses without authentication info?
> 
> Cheers,
> 



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Re: [asterisk-users] Change Media IP in SDP

[Max Grobecker]

Hi,

normally, Asterisk handles RTP IP addresses in SDP correctly, if you have 
specified
 - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia)
 - your local network with "localnet=yournetwork/networkmask" - e.g. 
"localnet=192.168.1.0/255.255.255.0"
 - directmedia, canreinvite, directrtpsetup is deacitivated

In this case, your Asterisk will always stay in the RTP stream and signalling 
only it's own IP address to other peers
which is configured for the interface which Asterisk uses to reach the peer.


But: If you have only one interface configured on your Asterisk server and an 
external firewall/router is managing your separated networks, this might not 
help.
In this case you can use "externip" on a per peer basis in your SIP 
configuration to specify the IP address Asterisk uses in the SDP.
Maybe, a global configuration of "externip" and "localnet" is all you need to 
help Asterisk setting the SDP address correctly.
Also, enabling ICE support can help you getting the correct IP address if the 
remote peer supports it.



Greetings
 Max


Am 07.12.2016 um 00:02 schrieb Harel:
> Hello List,
> I need your help with information going out on my SDP.
> Is it possible to update the Media Address on a per-call basis or a 
> per-channel basis?
> Reason:
> My Asterisk is in a private network and needs to connect to UA on its 
> internal network and also few external networks. One network is public and 
> the others are not public. Between each other the external networks are not 
> routable. Signaling is flowing with no issues because SIP Registers and NAT 
> boxes maintain sessions correctly. The problem is with RTP. After making 
> traces on all possible nodes of this network I clearly found out that the RTP 
> fails because the Asterisk doesn't manage to communicate the correct address 
> to the UAs in the SDP. It will report its internal IP address and the remote 
> UA will try to send its RTP to this address which, of course, will fail 
> miserably. 
> Obviously I can't use externaddr or media_address in sip.conf because it will 
> only be good for one network while the other external networks will fail just 
> the same. Same applies for STUN, it will only be good for the network the 
> STUN requests are being sent from. 
> On all networks I have fix IP addresses on my side and I fully control a 
> professional security box. 
> Asterisk is 13.6.0
> I can't, and don't want to, touch user-side equipment which is normally some 
> kind of voip phone behind a standard home VDSL router.
> 
> Any ideas how can I transmit the correct IP address in SDP to UAs on 
> different networks?
> 
> Many thanks,
> Harel
> 
> 



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[asterisk-users] Change Media IP in SDP

[Harel]

Hello List,
I need your help with information going out on my SDP.
Is it possible to update the Media Address on a per-call basis or a per-channel 
basis?
Reason:
My Asterisk is in a private network and needs to connect to UA on its internal 
network and also few external networks. One network is public and the others 
are not public. Between each other the external networks are not routable. 
Signaling is flowing with no issues because SIP Registers and NAT boxes 
maintain sessions correctly. The problem is with RTP. After making traces on 
all possible nodes of this network I clearly found out that the RTP fails 
because the Asterisk doesn't manage to communicate the correct address to the 
UAs in the SDP. It will report its internal IP address and the remote UA will 
try to send its RTP to this address which, of course, will fail miserably. 
Obviously I can't use externaddr or media_address in sip.conf because it will 
only be good for one network while the other external networks will fail just 
the same. Same applies for STUN, it will only be good for the network the STUN 
requests are being sent from. 
On all networks I have fix IP addresses on my side and I fully control a 
professional security box. 
Asterisk is 13.6.0
I can't, and don't want to, touch user-side equipment which is normally some 
kind of voip phone behind a standard home VDSL router.

Any ideas how can I transmit the correct IP address in SDP to UAs on different 
networks?

Many thanks,
Harel


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[asterisk-users] Asterisk install challenge

[christopher kamutumwa]

am new to asterisk and am trying to "make all". or dahdi install but in
only reach this stage below on centos 6.8 . Any idea how to resolve or
bypass this


configure: creating ./config.status
./configure: line 18858: cannot create temp file for here-document: No such
file or directory
configure: error: write failure creating ./config.status
make: *** [all] Error 1


Thanks,

Chris
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Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK

[FrancisM]

Yes its possible mine im using 2pcs of Huwawie 173 USB dongle (one for each
ISP provider) and connected to raspberry-pi. I can call out and recived
call with IVR to forward to the extensio numbers.


On Tue, 6 Dec 2016 at 10:36, christopher kamutumwa 
wrote:

> Is it possible to have a simcard configured and become incoming line
>
> and outgoing on asterisk and also have the IVR function? If yes wat
>
> hardware is required to have this Accompished
>
>
>
> Thanks
>
>
>
>
>
> Chris
>
>
>
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>
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> https://community.asterisk.org/
>
>
>
> New to Asterisk? Start here:
>
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
>
>
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Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK

[Frank Vanoni]

Hi Chris

On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote:

> Is it possible to have a simcard configured and become incoming line
> and outgoing on asterisk and also have the IVR function? 

Yes, it is possible! :-)

A cheap solution is using a 3G-UBS-dongle. 

I have two SIM cards working in my Asterisk. I'm using two 3G-dongles
(one for each SIM), a Huawei E173 (firmware 11.126.85.00.209) and a
Huawei E180 (firmware 11.126.10.01.68). 

Google for "asterisk chan dongle" and you will find plenty of infos.





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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Toshaan Bharvani | VanTosh]

On 05/12/16 17:57, Gopalakrishnan N wrote:

True agree, problem is somehow the people purchased am supporting to
overcome that. Trying level best... around 20 phones has been
purchased

I have been able to use the following Cisco IP Phone with Asterisk.
- Cisco SPA303, Cisco SPA504, Cisco 7941 and Cisco 8941
The SPA version just work as they are real SIP clients, however the IP 
range, 7xxx, 8xxx, 9xxx are SCCP clients, converting them as you did to 
SIP clients, is the Cisco SIP client and runs over TCP, not UDP as a 
normal SIP client.
Coming back to your question, you will need to setup a DHCP with BOOTP 
and TFTP wit XML distribution, as you have done for the firmware 
distribution and Asterisk needs to have SIP wit TCP support.
Additionnaly you need to generate a Cisco SIP XML configuration file and 
place it on your TFTP server with the MAC address.
The directory and video I have never been able to get working, however 
as a SIP client, it works fine.
I have these phones running with a number of customers in production and 
they all work, we actually mix types of phones, as not to require 
customers to purchase new IP phones, however sometimes the effort and 
time, which translates into cost, is too high.








On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, > wrote:

With all the money you plan to invest in firmware, licenses, etc.,
you have bought a Grandstream IP phone or Yealink...
Better still use a Raspberry Pi as a IP Softphone, or your Android phone 
with SIP client.




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