Re: [asterisk-users] 1000 analogue lines with asterisk

[Harry McGregor]

Hi,

For analog, I really like telco grade channel banks.

I would recommend the adit 600, there is a good market on Ebay, and you 
can do 48 channels per adit 600, with 2 T1 interfaces.  Having onsite 
spares would not be an issue (cost is low).  You can put two next to 
each other in a rack, taking up about 2U of space per 2 channel banks.


You could service this with six eight port T1 cards, or with 
eleven/twelve quad T1 cards.  I would distribute across two, three, or 
even four servers for redundancy/resiliency and load balancing.


-Harry

On 02/17/2016 12:16 AM, Goke Aruna wrote:


On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani > wrote:


Sangoma 50 port FXS 




Thanks.
Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
Regards




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Re: [asterisk-users] 1000 analogue lines with asterisk

[Goke Aruna]

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani  wrote:

> Sangoma 50 port FXS



Thanks.
Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
Regards
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Mitul Limbani]

Use Sangoma 50 port FXS
On Feb 17, 2016 12:42 PM, "Goke Aruna"  wrote:

> Thanks Mitul,
> The server spec is okay but I need information on the fxs hardware to use.
> Regards
>
> On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani  wrote:
>
>> Quad core Xeon with 4GB ram
>> On Feb 17, 2016 12:32 PM, "Goke Aruna"  wrote:
>>
>>> Hello all,
>>> Can someone recommend what hardware to use for a 1000 analogue line
>>> capacity asterisk PABX?
>>>
>>> Regards
>>>
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Goke Aruna]

Thanks Mitul,
The server spec is okay but I need information on the fxs hardware to use.
Regards

On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani  wrote:

> Quad core Xeon with 4GB ram
> On Feb 17, 2016 12:32 PM, "Goke Aruna"  wrote:
>
>> Hello all,
>> Can someone recommend what hardware to use for a 1000 analogue line
>> capacity asterisk PABX?
>>
>> Regards
>>
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Mitul Limbani]

Quad core Xeon with 4GB ram
On Feb 17, 2016 12:32 PM, "Goke Aruna"  wrote:

> Hello all,
> Can someone recommend what hardware to use for a 1000 analogue line
> capacity asterisk PABX?
>
> Regards
>
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[asterisk-users] 1000 analogue lines with asterisk

[Goke Aruna]

Hello all,
Can someone recommend what hardware to use for a 1000 analogue line
capacity asterisk PABX?

Regards
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Re: [asterisk-users] NAT on IPsec Tunnel

[Gopalakrishnan N]

Finally got it worked, the issue was E164 callerid format, where i set it
up, after removing the E164 format its was thru.

Regards

On Fri, Feb 12, 2016 at 9:31 PM Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Now incoming works fine, this is because of my SonicWALL firmware issue,
> tried with different SonicWALL inbound works.
>
> But for outbound am getting 408 request time out error in the NAT on VPN
> tunnel.
>
> On Fri, Feb 12, 2016 at 3:50 AM Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi all,
>>
>> Am using Asterisk 11.2.1. And for site testing, Verizon is doing Interop
>> testing with site to site IPsec tunnel and with public IP over the tunnel.
>>
>> Problem is when I do an inbound call, only a IVR message plays, whereas
>> am not able to transfer a call to extension, or dtmf not even works and
>> even outbound getting 408 request timeout.
>>
>> By all means I have configured externaddr and localnet in my sip.conf.
>>
>> Verizon says still the contact information shows my private IP, even
>> though I configured externaddr.
>>
>> When I route the incoming call directly to an hardphone extension, am not
>> able to answer the call in the hardphone, the ring LED keeps in blinking
>> even though i pickup the receiver, which is strange i haven;t seen.
>>
>> Can someone had any of this issue or throwing out any information would
>> help me.
>>
>> *Attached PCAP File:*
>> InboundCall_Direct_Extension - not able to answer in the hardphone
>> InboundCall_DTMF - Inbound call plays a message and wait for DTMF, where
>> its not recognized
>>
>> Tks.
>>
>
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Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

[Sonny Rajagopalan]

I can confirm that the server is receiving the SIP request, but simply
doesn't do anything with it (log from the server below). Does this have
anything to do with how PJSIP was compiled or configured?:

Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:11.12.13.14 SIP/2.0
Method: REGISTER
Request-URI: sip:11.12.13.14
Request-URI Host Part: 11.12.13.14
[Resent Packet: False]
Message Header
Via: SIP/2.0/TCP 192.168.1.16:54402
;rport;branch=z9hG4bKPjpaCDxnhZT22z-O183o5uZzMHNwTNrpkl;alias
Transport: TCP
Sent-by Address: 192.168.1.16
Sent-by port: 54402
RPort: rport
Branch: z9hG4bKPjpaCDxnhZT22z-O183o5uZzMHNwTNrpkl
alias
Route: 
Route URI: sip:11.12.13.14;transport=tcp;lr
Route Host Part: 11.12.13.14
Route URI parameter: transport=tcp
Route URI parameter: lr
Max-Forwards: 70
From: ;tag=Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
SIP from address: sip:987654321@11.12.13.14
SIP from address User Part: 987654321
SIP from address Host Part: 11.12.13.14
SIP from tag: Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
To: 
SIP to address: sip:987654321@11.12.13.14
SIP to address User Part: 987654321
SIP to address Host Part: 11.12.13.14
Call-ID: 8NDmEFaT2lmQRMUBf77UrRKRBIc3cT0h
CSeq: 29457 REGISTER
Sequence Number: 29457
Method: REGISTER
Supported: outbound, path
Contact: ;reg-id=1;+sip.instance=""
Contact URI: sip:987654321@192.168.1.16:54402;transport=TCP;ob
Contact URI User Part: 987654321
Contact URI Host Part: 192.168.1.16
Contact URI Host Port: 54402
Contact URI parameter: transport=TCP
Contact URI parameter: ob
Contact parameter: reg-id=1
Contact parameter:
+sip.instance=""\r\n
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


On Mon, Feb 15, 2016 at 6:01 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:

> Nope, there are no contacts to  show that pertain to these endpoints (only
> my SIP trunks show up).
>
> On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp  wrote:
>
>> Sonny Rajagopalan wrote:
>>
>>> Does this help:
>>>
>>
>> Yes, the transport parameter is in the Contact header so it's interesting
>> it didn't work. If you use pjsip show contacts what is the contact for the
>> AOR?
>>
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
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>
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Re: [asterisk-users] SIP URI set 'telephone-context='

[imperium broadcast]

Thanks for the reply Trey, should of said I'm using chan_sip.

Regards
Mick
On 16 Feb 2016 18:03, "Trey Hilyard"  wrote:

> Are you using res_pjsip or chan_sip?
>
> For PJSIP, it's as easy as passing the parameters to the Dial. For example:
> Dial(PJSIP/${ARG1}\;phone-context=mydomain.com@pjsippeer,60)
>
> I am pretty sure it was easy in chan_sip, too. If you are using chan_sip,
> I'll try and find an example.
>
> On Tue, Feb 16, 2016 at 11:03 AM imperium broadcast <
> imperium.broadc...@gmail.com> wrote:
>
>> Hi all, I am currently using asterisk 11, and I am trying to figure out
>> how to set the uri parameter telephone-context.
>> I need to set it for outbound calls for a specific carrier when making
>> emergency calls and don't seem able to find the option to set it.
>>
>> Regards
>> Impy
>> aka Mick
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Re: [asterisk-users] SIP URI set 'telephone-context='

[Trey Hilyard]

Are you using res_pjsip or chan_sip?

For PJSIP, it's as easy as passing the parameters to the Dial. For example:
Dial(PJSIP/${ARG1}\;phone-context=mydomain.com@pjsippeer,60)

I am pretty sure it was easy in chan_sip, too. If you are using chan_sip,
I'll try and find an example.

On Tue, Feb 16, 2016 at 11:03 AM imperium broadcast <
imperium.broadc...@gmail.com> wrote:

> Hi all, I am currently using asterisk 11, and I am trying to figure out
> how to set the uri parameter telephone-context.
> I need to set it for outbound calls for a specific carrier when making
> emergency calls and don't seem able to find the option to set it.
>
> Regards
> Impy
> aka Mick
> --
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[asterisk-users] SIP URI set 'telephone-context='

[imperium broadcast]

Hi all, I am currently using asterisk 11, and I am trying to figure out how
to set the uri parameter telephone-context.
I need to set it for outbound calls for a specific carrier when making
emergency calls and don't seem able to find the option to set it.

Regards
Impy
aka Mick
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Re: [asterisk-users] Voicemail using object storage?

[Olivier]

Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be
mounted on Linux and thus allow any application like Asterisk make use of
it ?

[1] https://github.com/s3fs-fuse/s3fs-fuse

2016-02-16 1:05 GMT+01:00 Andrew Ruthven :

> Hey,
>
> I've found a bit of chatter about people using hacks to copy voicemail
> messages into object storage (like S3) after they've been recorded. But
> I was wondering if any work has been done on the VoiceMail app to
> actually store and retrieve messages to/from an object store?
>
> Cheers,
> Andrew
> --
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> MIITP, ITCP
>
> At work: andrew.ruth...@catalyst.net.nz
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>
>
>
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Re: [asterisk-users] Best place to issue tickets for Digium phones ?

[Olivier]

Thanks for replying.

I'll try to open up a ticket there.

2016-02-11 16:40 GMT+01:00 Shaun Ruffell :

> On Wed, Feb 10, 2016 at 03:11:02PM +0100, Olivier wrote:
> > Hello,
> >
> > I've recently given a try to a Digium D70 phone.
> >
> > At the moment, I'm configuring them though config files with a DHCP
> server
> > and not using DPMA.
> > Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0).
> >
> >
> > Which is the best place to:
> > - read about past issues
> > - open new tickets for remaining issues.
> >
> > Best regards
>
> Hi Olivier,
>
> I would recommend checking out  http://support.digium.com for
> assistence with any of Digium's commercial products, including the
> phones.
>
> If you have an issue you can open up support tickets through that
> site as well.
>
> Cheers,
> Shaun
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
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