I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't call PSTN (call goes off when I pick the phone up). What can I do? 2)What is [EMAIL PROTECTED] ? For what is used? 3)Can I transfer calls? I guess that if I do transfer = yes in the general section of sip.conf, it should work... but it doesn't!! 4)And finally, the caller ID. I have done usecallerid=yes in the general section of sip.conf and the I put callerid="SIP" <2222> in the [sip] section (the one that I have created for my devide). But it doesn't work either! Any ideas? My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid="sip" <2222> username=sip host=188.208.12.37 accountcode=sip Thanks you all!!! Michelle ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/
Hi Michelle,
For the d-link VoIP gateways you need to configure your mgcp.conf the
following is my configuration for a dg104s.
Mgcp.conf
--------------
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 0.0.0.0
[mgcp01]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
host = dynamic
context = sipstart
callerid = Edwin Silva <6014>
line => aaln/4
callerid = Edwin Silva <6013>
line => aaln/3
callerid = Edwin Silva <6012>
line => aaln/2
callerid = Edwin Silva <6011>
line => aaln/1
nat=1
And the following is my entry in extensions.conf
Extensions.conf
-----------------
;MGCP Phones (DG-104S)
exten => _60XX,1,Dial,MGCP/aaln/${EXTEN:3}@mgcp0${EXTEN:2:1}|20
exten => _60XX,2,Congestion
exten => _60XX,102,Congestion
This should let you call any device on the dg102s without any problems
just make sure that you configure your dg-104s to send out mgcp01 or
whatever you decide to use as the entry in your mgcp.conf and change the
pattern matching in extensions.conf to reflect this as well.
Edwin Silva
WW Works Inc.
3060 Mainway Dr. Unit 104
Burlington, ON
L7M 1A3
(905) 332-5844 ext. 517
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of michelle
matis litio
Sent: Wednesday, June 11, 2003 3:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] some sip questions AGAIN
I write the email again, the third time!!, cause the other two ones, I
have
had problems while sending them. I hope this time it works. Here is the
email again:
Hi (and sorry) everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly
ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN
above all, but I have two dlink dg102s (MGCP) and I'd like to can call
them
too. The problem is that when I use g723 I can call MGCP but I can't
call
PSTN (call goes off when I pick the phone up). What can I do?
2)What is [EMAIL PROTECTED] ? For what is used?
3)Can I transfer calls? I guess that if I do transfer = yes in the
general
section of sip.conf, it should work... but it doesn't!!
4)And finally, the caller ID. I have done usecallerid=yes in the general
section of sip.conf and the I put callerid="SIP" <2222> in the
[sip]
section
(the one that I have created for my devide). But it doesn't work either!
Any
ideas?
My sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
transfer = yes
threewaycalling = yes
usecallerid = yes
hidecallerid = no
[sip]
type=friend
callerid="sip" <2222>
username=sip
host=188.208.12.37
accountcode=sip
Thanks you all!!!
Michelle
-----
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Hi Edwin I have my mgcp.conf almost the same as yours, except from "nat=1" , why do you put it? Anyway, DL102s now works more or less acceptably so now I'm having a battle with sip.conf !!!! Thank you for your help Michelle ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/
Hi Edwin! (and everybody) I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the phones conected to the SIP GW and the ones conected to the MGCP GW goes OK ONLY if I call from the MGCP to the SIP. Phones at MGCP can call without problems to the PSTN (voice quality isn't very good, with silence times, but it can be supported!). But phones at SIP can't do any call! The problem is that when I pick up the callee phone, I don't hear nothing and the call goes off inbetween 4 or 5 seconds. And the caller (SIP) doesn't realise I have picked up, because It's still hearing the calling tone.When the call goes off, the caller hear the congestion tone. I don't know what is the problem!!!! I can't achive to transfer calls. When I dial #, it doesn't happen anything!! And the callerID doesn't work either. My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid="sip" <2222> username=sip host=188.208.12.37 accountcode=sip My extensions.conf exten => 2222,1,dial,SIP/2222@188.208.12.37|60|rTt exten => 2222,2,Hangup Thanks very much for any help!!! Bye Michelle>Nat=1 is so that mgcp functions properly behind a NAT gateway. >What kind of problems are you having with your SIP? What type of SIP >phone do you have? Can you elaborate a little more or even post you >SIP.conf? >Here's what ours looks like so you can do a comparison: >Sip.conf >----------- >; >; SIP Configuration for Asterisk >; >[general] >port = 5060 ; Port to bind to >bindaddr = 0.0.0.0 ;Address to bind to >context = sipstart ; Default for incoming calls>tos = lowdelay >[sip_phone] >type=friend >username=sip_phone >secret=sip_phone >host=dynamic >nat=1 >-----Original Message----- >From: href="javascript:sendMsg('asterisk-users- asterisk-users- admin@lists.digium.com');">admin@lists.digium.com');">asterisk-users- admin@lists.digium.com>[mailto:asterisk-users-admin@lists.digium.com]');">admin@lists.digium.com]');">[mailto:asterisk- users-admin@lists.digium.com] On Behalf Of michelle >matis litio >Sent: Wednesday, June 11, 2003 12:12 PM >To: asterisk- users@lists.digium.com');">users@lists.digium.com');">asterisk- users@lists.digium.com>Subject: [Asterisk-Users] Re:Some SIP questions AGAIN >Hi Edwin >I have my mgcp.conf almost the same as yours, except from "nat=1" ,why >do you put it? >Anyway, DL102s now works more or less acceptably so now I'm having a >battle with sip.conf !!!! >Thank you for your help >Michelle >----- >Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com/app/message?l=es&o=8&url=http% 3A%2F%2Fmixmail%2Eya%2Ecom" target=_blank>http://mixmail.ya.com Ya.com ADSL >Home 24 h, Módem + Alta ¡Gratis! href="http://mixmail.ya.com/app/message?l=es&o=8&url=http%3A% 2F%2Facceso%2Eya%2Ecom%2Fadslhome24h%2F" target=_blank>http://acceso.ya.com/adslhome24h/>_______________________________________________ >Asterisk-Users mailing list >Asterisk- Users@lists.digium.com');">Users@lists.digium.com');">Asterisk- Users@lists.digium.com > href="http://mixmail.ya.com/app/message? l=es&o=8&url=http%3A% 2F%2Flists%2Edigium%2Ecom%2Fmailman%2Flistinfo%2Fasterisk% 2Dusers" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk- users >_______________________________________________>Asterisk-Users mailing list >http://lists.digium.com/mailman/listinfo/asterisk-');">Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk- Asterisk-Users@lists.digium.com');">users');">Asterisk- Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk-users----- ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/
Hi everybody one more time!
I also have done a SIP debug and that's an extract of what I have found:
(...)
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000
to 229.159.241.112:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7-
8c6b606-10eb
From: <sip:sip@188.208.12.37:5060> ;tag=0-13c4-3a5246f7-8c6b604-c3a
To: <sip:3333@188.208.12.237>;tag=as52ed0a6a
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:3333@188.208.12.237>
Content-Type: application/sdp
Content-Length: 135
v=0
o=root 11673 11673 IN IP4 188.208.12.237
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000
to 229.159.241.112:5060
-- Hungup 'IAX2[test]/1'
== Spawn extension (default, 3333, 1) exited non-zero
on 'SIP/229.159.241.112:5
060'
set_destination: Parsing <sip:sip@188.208.12.37:5060> for address/port to
send t
o
set_destination: set destination to 188.208.12.37, port 5060
Reliably Transmitting:
BYE sip:sip@188.208.12.37:5060 SIP/2.0
Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d
From: <sip:3333@188.208.12.237>;tag=as52ed0a6a
To: <sip:sip@188.208.12.37:5060> ;tag=0-13c4-3a5246f7-8c6b604-c3a
Contact: <sip:3333@188.208.12.237>
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 188.208.12.37:5060
Sip read:
SIP/2.0 200 OK
From: <sip:3333@188.208.12.237>
To: <sip:sip@188.208.12.37:5060> ;tag=0-13c4-3a5246f7-8c6b604-c3a
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37
CSeq: 102 BYE
Via: SIP/2.0/UDP
188.208.12.237:5060 ;received=188.208.12.237 ;branch=z9hG4bK67231
48d
Content-Length:0
7 headers, 0 lines
Message is BYE
I can't understand why the "out of SIP" messages go to an IP so
strange!!!
(229...)
Any ideas?
I've just sent my sip.conf and all in the previous message. Hope someone
can help!!
greetings
michelle
PD:188.208.12.237 is the asterisk IP
>>Michelle wrote:
Hi Edwin! (and everybody)
I have some questions about SIP, as I wrote in another mail. I have a SIP
Gateway and I have two phones conected to it.Also, I have two Dlink
dg102s with four phones conected to them. The main problems are two.
Calls between the phones conected to the SIP GW and the ones conected
to the MGCP GW goes OK ONLY if I call from the MGCP to the SIP. Phones
at MGCP can call without problems to the PSTN (voice quality isn't very
good, with silence times, but it can be supported!). But phones at SIP can't
do any call! The problem is that when I pick up the callee phone, I don't
hear nothing and the call goes off inbetween 4 or 5 seconds. And the
caller (SIP) doesn't realise I have picked up, because It's still
hearing the
calling tone.When the call goes off, the caller hear the congestion tone. I
don't know what is the problem!!!!
I can't achive to transfer calls. When I dial #, it doesn't happen
anything!!
And the callerID doesn't work either.
My sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
transfer = yes
threewaycalling = yes
usecallerid = yes
hidecallerid = no
[sip]
type=friend
callerid="sip" <2222>
username=sip
host=188.208.12.37
accountcode=sip
My extensions.conf
exten => 2222,1,dial,SIP/2222@188.208.12.37|60|rTt
exten => 2222,2,Hangup
Thanks very much for any help!!!
Bye
Michelle
;-----Original Message-----
>From: <A href="javascript:sendMsg('asterisk-users-
admin@lists.digium.com');">asterisk-users-admin@lists.digium.com</A>
><A href="javascript:sendMsg('[mailto:asterisk-users-
admin@lists.digium.com]');">[mailto:asterisk-users-admin@lists.digium.com]
</A> On Behalf Of michelle >matis litio >Sent: Wednesday,
June 11,
2003 12:12 PM >To: <A href="javascript:sendMsg('asterisk-
users@lists.digium.com');">asterisk-users@lists.digium.com</A>
>Subject: [Asterisk-Users] Re:Some SIP questions AGAIN >Hi Edwin
>I have my mgcp.conf almost the same as yours, except from
"nat=1" ,
why >do you put it? >Anyway, DL102s now works more or less
acceptably so now I'm having a >battle with sip.conf !!!!
>Thank you
for your help >Michelle >----- >Tu cuenta de correo
gratuita Mixmail
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