hi all,
how to establish a call between two asterisk servers for the sip users
registered for the servers.
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, February 10, 2008 11:30 PM
Subject: asterisk-users Digest, Vol 43, Issue 30
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> Today's Topics:
>
> 1. Re: Domainname for outgoing uri-dialing (B. Haje)
> 2. Re: oneway audio with asterisk behind cisco pix 506 (Adam KOSA)
> 3. Re: Asterisk Scalability (Bryan M. Johns)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sun, 10 Feb 2008 18:11:01 +0100
> From: "B. Haje" <bhaje at gmx.de>
> Subject: Re: [asterisk-users] Domainname for outgoing uri-dialing
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <005501c86c07$edd74c60$141b0c0a at bjoern>
> Content-Type: text/plain; charset="us-ascii"
>
> asterisk-users-bounces at lists.digium.com wrote:
>> 8 feb 2008 kl. 13.24 skrev Bjoern Haje:
>>
>>> Hi,
>>>
>>> I use outgoing URI-dialing for my sip-phones as suggested in
>>> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
>>>
>>> The relevant extensions look like this:
>>>
>>> [dial-uri]
>>> exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>> exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>> exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>>
>>> [macro-uridial]
>>> exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)})
>>> exten =>
s,n,Set(CALLERID(number)=${CALLERID(number)}@domain.com)
>>>
>>> exten => s,n,Dial(SIP/${dialuri},120,tr)
>>> exten => s,n,Congestion()
>>>
>>> I end up with an outgoing SIP-Invite with contact and from-headers
>>> like username at domain.com@<IP-address>
>>>
>>> That obviously is not what I want. I can set the fromdomain value
in
>>> the general-part of my sip.conf and leave away the setting of the
>>> callerid which fixes the problem. But as I want to use different
>>> domains for the outgoing calls depending on the user, that is not a
>>> solution for me. Can I influence the generation of the outgoing
>>> domainname somehow?
>>
>> No, but that would be a good addition to Asterisk. I started
>> experimenting with that in my caller ID utf8 branch at some point,
>> but never got time or funding to complete that work.
>
> Thanks for your help again. Would be nice really, but I'll try to find
a
> workaround to avoid that problem (or ignore it).
>
> Bjoern
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Sun, 10 Feb 2008 18:44:46 +0100
> From: Adam KOSA <adamk at 3a.hu>
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
> pix 506
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <47AF380E.9090109 at 3a.hu>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>> permit udp any host 192.168.5.0 range 10000 20000 and then I didn't
>
> home users typically use /24 netmask. If this is the case, i don't
> understand why do you write keyword host following a network address.
>
> either specify a valid host address, or write 192.168.5.0 255.255.255.0
> to specify the whole subnet.
>
> if the netmask isn't /24 then, of course the above 5.0 may be a valid
> host address.
>
> regards
> adam
>
>
>
> ------------------------------
>
> Message: 3
> Date: Sun, 10 Feb 2008 12:54:44 -0500
> From: "Bryan M. Johns" <bryan at sheltonjohns.com>
> Subject: Re: [asterisk-users] Asterisk Scalability
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <78A8A577-3DDF-486C-A634-D17147A36CBF at
sheltonjohns.com>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> We have multiple installs that tested-out at nearly concurrent 400 SIP
> channels on a Dell 2950 with 2Xquad core at 1.6 Ghz, 16 GB of RAM.
>
> Bryan M. Johns
> Shelton | Johns
> Office: 678.248.2637
> FindMe: 678.229.1809
> Support: support at sheltonjohns.com
> http://www.sheltonjohns.com
>
> On Feb 8, 2008, at 5:09 AM, Femi wrote:
>
>> Hi,
>> Does anyone have data on the switching capacity of Asterisk based on
>> the
>> hardware?
>> I need to know what type of hardware would be required to switch 100
>> simultaneous calls as opposed to 1000 or 10000 calls, no TDM just
>> SIP to SIP
>> VoIP calls
>>
>> Thanks
>>
>> Femi
>>
>>
>>
>>
>>
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