20 Apr
2004
20 Apr
'04
8:03 p.m.
Hello Laurent,
Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
LB> I think i follow the instructions ... but should i "route" ( so modify
LB> the ser.cfg file) some messages to my gateway to use the answering
LB> machine or services provided by my gateway ...
LB> And what about autocreatepeer=yes?
surely later you will have to route in ser.cfg for outbound calls, checking first for
user auth and grp assignement.
Using the extention method you can direct sip addresses to services in
asterisk like voicemail or voice prompts.
that's my ser routing for outside calls
record_route();
if (uri=~"sip:0[0-9]+@"){
if (!proxy_authorize("mydomain.com",
"subscriber")) {proxy_challenge("mydomain.com",
"0");sl_send_reply("403", "That's not your
home");break;}; #fine proxy challenge
if (!is_user_in("credentials",
"local")){sl_send_reply("403", "No permission for local
calls");break;}; #fine invite
rewritehostport("sip.mydomain.com:5090");
t_relay();
break;
}; #fine if uri sip:0
inside asterisk calls forwarded by ser are treated by this extention
exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
exten => _0.,2,Congestion
its important to notice that you have to block port 5090 for incoming
ip requests ....
Only ser will be allowed to talk to asterisk and forward calls, after
user checking.
Hope it helps
LB> thx
LB> Alessio Focardi wrote:
>Hello Laurent,
>
>Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
>
>LB> But my clients should register on SER or on Asterisk?
>
>On ser, then you will need to protect asterisk from unallowed pstn
>call, but that will come later on.
>
>
>LB> thx
>
>LB> Alessio Focardi wrote:
>
>
>
>>>Hello Laurent,
>>>
>>>Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>>
>>>LB> In fact, my ser installation works fine...
>>>LB> I can pass call through asterisk in standalone...
>>>LB> The problem is to interconnect the 2, to register ( i don't know if
it's
>>>LB> a right solution) the sipphones on SER and to go outside thanks to
>>>LB> asterisk...
>>>
>>>make asterisk use port 5090 for sip, then as a first step make
>>>asterisk register in ser as an extention.
>>>
>>>you can do this in asterisk's sip.conf
>>>
>>>example
>>>
>>>register => 10:password@sip.yourdomanin.com/10
>>>
>>>this tells asterisk to register extention 10 as address 109(a)yourdomain.com
>>>
>>>dial 10 with a sip phone and you are in asterisk ... note that you
>>>should have an extention 10 defined, or it will not work.
>>>
>>>B> Thx
>>>
>>>LB> Alessio Focardi wrote:
>>>
>>>
>>>
>>>
>>>
>>>>>Hello Laurent,
>>>>>
>>>>>What you want to accomplish could be done, my advice is to setup a
>>>>>working installation of ser then you will continue with asterisk.
>>>>>
>>>>>At first make a simple installation of ser (no auth, no db maybe) and
make
>>>>>your phones call each other.
>>>>>
>>>>>If you encounter specific problems and you want to have some help
this
>>>>>is the right place.
>>>>>
>>>>>Good luck !
>>>>>
>>>>>
>>>>>
>>>>>Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>>
>>>>>LB> Hi,
>>>>>LB> I want to use SER as a sip Proxy and asterisk as a gateway to
the PSTN
>>>>>LB> network ...
>>>>>LB> My sipphones are BudgetTone101 and i'm having trouble
trying configure
>>>>>LB> them....
>>>>>LB> Indeed, i don't know if they should register on SER or
not... I don't
>>>>>LB> know what kind of sip messages should be passed to my machine
running
>>>>>LB> asterisk.
>>>>>LB> I don't know what must be in ser.cfg ( if you've an
example it could
>>>>>LB> help me a lot...)...
>>>>>LB> I wasn't able to find documentations about using Ser and
Asterisk in
>>>>>LB> this configuration ( messages in the archives are not explicit
>>>>>LB> enough....) , so if you've a pointer or so....
>>>>>
>>>>>
>>>>>LB> Help...
>>>>>
>>>>>LB> thx,
>>>>>LB> Laurent
>>>>>
>>>>>LB> _______________________________________________
>>>>>LB> Serusers mailing list
>>>>>LB> serusers(a)lists.iptel.org
>>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>LB> _______________________________________________
>>>LB> Serusers mailing list
>>>LB> serusers(a)lists.iptel.org
>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>
>
>
>
>
--
Best regards,
Alessio mailto:afoc@interconnessioni.it
23 Apr
23 Apr
12:21 p.m.
New subject: [Serusers] SER + Asterisk
Hi,
First of all, thanks a lot Alessio for your help ... I m not so far away
from the end of my quest....
I m still encountering one problem.... When i pass call to the pstn from
ser through asterisk, even if it succeeds, the value on my client is
still 404... My mobile phone is ringing but i see on the Xlite screen
404, i don't understand ...
thx
Laurent
Alessio Focardi wrote:
Hello Laurent,
Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
LB> I think i follow the instructions ... but should i "route" ( so modify
LB> the ser.cfg file) some messages to my gateway to use the answering
LB> machine or services provided by my gateway ...
LB> And what about autocreatepeer=yes?
surely later you will have to route in ser.cfg for outbound calls, checking first for
user auth and grp assignement.
Using the extention method you can direct sip addresses to services in
asterisk like voicemail or voice prompts.
that's my ser routing for outside calls
record_route();
if (uri=~"sip:0[0-9]+@"){
if (!proxy_authorize("mydomain.com",
"subscriber")) {proxy_challenge("mydomain.com",
"0");sl_send_reply("403", "That's not your
home");break;}; #fine proxy challenge
if (!is_user_in("credentials",
"local")){sl_send_reply("403", "No permission for local
calls");break;}; #fine invite
rewritehostport("sip.mydomain.com:5090");
t_relay();
break;
}; #fine if uri sip:0
inside asterisk calls forwarded by ser are treated by this extention
exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
exten => _0.,2,Congestion
its important to notice that you have to block port 5090 for incoming
ip requests ....
Only ser will be allowed to talk to asterisk and forward calls, after
user checking.
Hope it helps
LB> thx
LB> Alessio Focardi wrote:
>Hello Laurent,
>
>Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
>
>LB> But my clients should register on SER or on Asterisk?
>
>On ser, then you will need to protect asterisk from unallowed pstn
>call, but that will come later on.
>
>
>LB> thx
>
>LB> Alessio Focardi wrote:
>
>
>
>
>
>>>Hello Laurent,
>>>
>>>Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>>
>>>LB> In fact, my ser installation works fine...
>>>LB> I can pass call through asterisk in standalone...
>>>LB> The problem is to interconnect the 2, to register ( i don't know if
it's
>>>LB> a right solution) the sipphones on SER and to go outside thanks to
>>>LB> asterisk...
>>>
>>>make asterisk use port 5090 for sip, then as a first step make
>>>asterisk register in ser as an extention.
>>>
>>>you can do this in asterisk's sip.conf
>>>
>>>example
>>>
>>>register => 10:password@sip.yourdomanin.com/10
>>>
>>>this tells asterisk to register extention 10 as address 109(a)yourdomain.com
>>>
>>>dial 10 with a sip phone and you are in asterisk ... note that you
>>>should have an extention 10 defined, or it will not work.
>>>
>>>B> Thx
>>>
>>>LB> Alessio Focardi wrote:
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>>>Hello Laurent,
>>>>>
>>>>>What you want to accomplish could be done, my advice is to setup a
>>>>>working installation of ser then you will continue with asterisk.
>>>>>
>>>>>At first make a simple installation of ser (no auth, no db maybe) and
make
>>>>>your phones call each other.
>>>>>
>>>>>If you encounter specific problems and you want to have some help
this
>>>>>is the right place.
>>>>>
>>>>>Good luck !
>>>>>
>>>>>
>>>>>
>>>>>Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>>
>>>>>LB> Hi,
>>>>>LB> I want to use SER as a sip Proxy and asterisk as a gateway to
the PSTN
>>>>>LB> network ...
>>>>>LB> My sipphones are BudgetTone101 and i'm having trouble
trying configure
>>>>>LB> them....
>>>>>LB> Indeed, i don't know if they should register on SER or
not... I don't
>>>>>LB> know what kind of sip messages should be passed to my machine
running
>>>>>LB> asterisk.
>>>>>LB> I don't know what must be in ser.cfg ( if you've an
example it could
>>>>>LB> help me a lot...)...
>>>>>LB> I wasn't able to find documentations about using Ser and
Asterisk in
>>>>>LB> this configuration ( messages in the archives are not explicit
>>>>>LB> enough....) , so if you've a pointer or so....
>>>>>
>>>>>
>>>>>LB> Help...
>>>>>
>>>>>LB> thx,
>>>>>LB> Laurent
>>>>>
>>>>>LB> _______________________________________________
>>>>>LB> Serusers mailing list
>>>>>LB> serusers(a)lists.iptel.org
>>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>LB> _______________________________________________
>>>LB> Serusers mailing list
>>>LB> serusers(a)lists.iptel.org
>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>
>
>
>
>
12:34 p.m.
New subject: [Serusers] SER + Asterisk
I answer to myself , i forgot a break; after the rewritehostport ...
Sorry,
Laurent
Laurent BURGY wrote:
Hi,
First of all, thanks a lot Alessio for your help ... I m not so far
away from the end of my quest....
I m still encountering one problem.... When i pass call to the pstn
from ser through asterisk, even if it succeeds, the value on my client
is still 404... My mobile phone is ringing but i see on the Xlite
screen 404, i don't understand ...
thx
Laurent
Alessio Focardi wrote:
Hello Laurent,
Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
LB> I think i follow the instructions ... but should i "route" ( so
modify
LB> the ser.cfg file) some messages to my gateway to use the answering
LB> machine or services provided by my gateway ...
LB> And what about autocreatepeer=yes?
surely later you will have to route in ser.cfg for outbound calls,
checking first for
user auth and grp assignement.
Using the extention method you can direct sip addresses to services in
asterisk like voicemail or voice prompts.
that's my ser routing for outside calls
record_route();
if (uri=~"sip:0[0-9]+@"){
if (!proxy_authorize("mydomain.com", "subscriber"))
{proxy_challenge("mydomain.com", "0");sl_send_reply("403",
"That's
not your home");break;}; #fine proxy challenge
if (!is_user_in("credentials",
"local")){sl_send_reply("403", "No permission for local
calls");break;}; #fine invite
rewritehostport("sip.mydomain.com:5090");
t_relay();
break;
}; #fine if uri sip:0
inside asterisk calls forwarded by ser are treated by this extention
exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
exten => _0.,2,Congestion
its important to notice that you have to block port 5090 for incoming
ip requests ....
Only ser will be allowed to talk to asterisk and forward calls, after
user checking.
Hope it helps
LB> thx
LB> Alessio Focardi wrote:
Hello
Laurent,
Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
LB> But my clients should register on SER or on Asterisk?
On ser, then you will need to protect asterisk from unallowed pstn
call, but that will come later on.
LB> thx
LB> Alessio Focardi wrote:
>> Hello Laurent,
>>
>> Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>
>> LB> In fact, my ser installation works fine...
>> LB> I can pass call through asterisk in standalone...
>> LB> The problem is to interconnect the 2, to register ( i don't
>> know if it's
>> LB> a right solution) the sipphones on SER and to go outside
>> thanks to
>> LB> asterisk...
>>
>> make asterisk use port 5090 for sip, then as a first step make
>> asterisk register in ser as an extention.
>>
>> you can do this in asterisk's sip.conf
>>
>> example
>>
>> register => 10:password@sip.yourdomanin.com/10
>>
>> this tells asterisk to register extention 10 as address
>> 109(a)yourdomain.com
>>
>> dial 10 with a sip phone and you are in asterisk ... note that you
>> should have an extention 10 defined, or it will not work.
>>
>> B> Thx
>>
>> LB> Alessio Focardi wrote:
>>
>>
>>
>>
>>
>>
>>>> Hello Laurent,
>>>>
>>>> What you want to accomplish could be done, my advice is to setup a
>>>> working installation of ser then you will continue with asterisk.
>>>>
>>>> At first make a simple installation of ser (no auth, no db
>>>> maybe) and make
>>>> your phones call each other.
>>>>
>>>> If you encounter specific problems and you want to have some
>>>> help this
>>>> is the right place.
>>>>
>>>> Good luck !
>>>>
>>>>
>>>>
>>>> Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>
>>>> LB> Hi,
>>>> LB> I want to use SER as a sip Proxy and asterisk as a gateway
>>>> to the PSTN
>>>> LB> network ...
>>>> LB> My sipphones are BudgetTone101 and i'm having trouble
>>>> trying configure
>>>> LB> them....
>>>> LB> Indeed, i don't know if they should register on SER or
>>>> not... I don't
>>>> LB> know what kind of sip messages should be passed to my
>>>> machine running
>>>> LB> asterisk.
>>>> LB> I don't know what must be in ser.cfg ( if you've an
>>>> example it could
>>>> LB> help me a lot...)...
>>>> LB> I wasn't able to find documentations about using Ser and
>>>> Asterisk in
>>>> LB> this configuration ( messages in the archives are not explicit
>>>> LB> enough....) , so if you've a pointer or so....
>>>>
>>>>
>>>> LB> Help...
>>>>
>>>> LB> thx,
>>>> LB> Laurent
>>>>
>>>> LB> _______________________________________________
>>>> LB> Serusers mailing list
>>>> LB> serusers(a)lists.iptel.org
>>>> LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>> LB> _______________________________________________
>> LB> Serusers mailing list
>> LB> serusers(a)lists.iptel.org
>> LB> http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>>
>>
>>
>>
>
7518
days inactive
7521
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2 comments
2 participants
participants (2)
-
Alessio Focardi -
Laurent BURGY