[Asterisk-Users] sip.conf
HI, can somebody tell me how and where must I put the SIP register line? I think is in [general] section of the sip.conf and that I have to put: register = user:[EMAIL PROTECTED]:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The same SIP problems...SORRY!
Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) 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 The other problem is that 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 username=sip host=188.208.12.37 accountcode=sip My extensions.conf exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten = ,2,Hangup I also have done a SIP debug and I'm sneding an extract of what I have found. I can't understand why the out of SIP messages go to an IP so strange!!! (229...) I can't find this IP anywhre in my system...Any ideas? Hope someone can help!! Thanks in advance! michelle PD:188.208.12.237 is the asterisk IP (...) 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: ;tag=0-13c4-3a5246f7-8c6b604-c3a To: ;tag=as52ed0a6a Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: 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, , 1) exited non-zero on 'SIP/229.159.241.112:5 060' set_destination: Parsing for address/port to send t o set_destination: set destination to 188.208.12.37, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d From: ;tag=as52ed0a6a To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Contact: Call-ID: [EMAIL PROTECTED] 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: To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Call-ID: [EMAIL PROTECTED] 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 - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] The same SIP problems...SORRY!
Hi!
I thought it was the SIP device too, but I have looked for avery litle comand
of this device and I can't find this Ip address, and I see that its Ip is Ok,
and I have configurated the REGISTRAR section too... I don't know what's
happening, and I don't understand that, if the IP is wrong, why can I hear
the callee phone ringing and the call only goes off when I pick it up?
it's so strange...I think!
Michelle
gt;On Mon, 16 Jun 2003, michelle matis litio wrote: gt;gt; to
229.159.241.112:5060 gt;gt; Retransmitting #5 (no NAT): gt;gt; SIP/2.0
200 OK gt;gt; Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-
3a5246f7- gt;gt; 8c6b606-10eb gt;gt; From: ;tag=0-13c4-3a5246f7-
8c6b604-c3a gt;gt; To: ;tag=as52ed0a6a gt;gt; Call-ID: A
href=javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]');f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]/A gt;gt; CSeq: 1 INVITE gt;gt; User-Agent:
Asterisk PBX gt;gt; Contact: gt;gt; Content-Type: application/sdp
gt;gt; Content-Length: 135 gt;gt; gt;gt; v=0 gt;gt; o=root 11673
11673 IN IP4 188.208.12.237 gt;gt; s=session gt;gt; c=IN IP4
188.208.12.237 gt;gt; t=0 0 gt;gt; =audio 13532 RTP/AVP 0 gt;gt;
a=rtpmap:0 PCMU/8000 gt;Hi, gt;Its being sent to that IP address,
because that is that the gt;originating SIP device put in its Via header.
gt;Also, your SIP device didn't put any From or To in its INVITE.
gt;Perhaps you could send a sip debug from the start of a SIP call
gt;attempt. gt;But I'm sure that the trouble is with your SIP Gateway
device's gt;setup. gt;Steve
gt;___ gt;Asterisk-
Users mailing list gt;A href=javascript:sendMsg('Asterisk-
[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-
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gt;http://lists.digium.com/mailman/listinfo/asterisk-users/A
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[Asterisk-Users] SIP REGISTER
Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: 501 Not impelmented back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1:UnRegisteredto: registrar: 188.208.12.237 5060expires: 2000 name: gateway passwd: 123 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no register = gateway:[EMAIL PROTECTED]/ [gateway] type=friend callerid=sip username=gateway host=188.208.12.37 secret=123 My extensions.conf exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten = ,2,Hangup I'm going crazy with this...I think that I'm not doing well the registration but I can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 is the IP of the SIP gateway. is one of the phones of the SIP Gateway...Anyone can helpPlease! Thanks very very much Michelle - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Re:Some SIP questions AGAIN
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
username=sip
host=188.208.12.37
accountcode=sip
My extensions.conf
exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt
exten = ,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-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');asterisk-users-
[EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');[mailto:asterisk-
[EMAIL PROTECTED]
On Behalf Of michelle matis litio Sent: Wednesday, June 11,
2003 12:12 PM To: asterisk-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');asterisk-
[EMAIL PROTECTED]
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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3A%2F%2Fmixmail%2Eya%2Ecom target=_blankhttp://mixmail.ya.com
Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis!
href=http://mixmail.ya.com/app/message?l=eso=8url=http%3A%
2F%2Facceso%2Eya%2Ecom%2Fadslhome24h%2F
target=_blankhttp://acceso.ya.com/adslhome24h/
___ Asterisk-
Users mailing list Asterisk-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');Asterisk-
[EMAIL PROTECTED] href=http://mixmail.ya.com/app/message?
l=eso=8url=http%3A%
2F%2Flists%2Edigium%2Ecom%2Fmailman%2Flistinfo%2Fasterisk%
2Dusers target=_blankhttp://lists.digium.com/mailman/listinfo/asterisk-
users ___
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-
Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com
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Re: RE: [Asterisk-Users] Re:Some SIP questions AGAIN
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:[EMAIL PROTECTED]:5060 ;tag=0-13c4-3a5246f7-8c6b604-c3a
To: sip:[EMAIL PROTECTED];tag=as52ed0a6a
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: sip:[EMAIL PROTECTED]
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, , 1) exited non-zero
on 'SIP/229.159.241.112:5
060'
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to
send t
o
set_destination: set destination to 188.208.12.37, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d
From: sip:[EMAIL PROTECTED];tag=as52ed0a6a
To: sip:[EMAIL PROTECTED]:5060 ;tag=0-13c4-3a5246f7-8c6b604-c3a
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]:5060 ;tag=0-13c4-3a5246f7-8c6b604-c3a
Call-ID: [EMAIL PROTECTED]
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
username=sip
host=188.208.12.37
accountcode=sip
My extensions.conf
exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt
exten = ,2,Hangup
Thanks very much for any help!!!
Bye
Michelle
;-Original Message-
gt;From: A href=javascript:sendMsg('asterisk-users-
[EMAIL PROTECTED]');[EMAIL PROTECTED]/A
gt;A href=javascript:sendMsg('[mailto:asterisk-users-
[EMAIL PROTECTED]');[mailto:[EMAIL PROTECTED]
/A On Behalf Of michelle gt;matis litio gt;Sent: Wednesday, June 11,
2003 12:12 PM gt;To: A href=javascript:sendMsg('asterisk-
[EMAIL PROTECTED]');[EMAIL PROTECTED]/A
gt;Subject: [Asterisk-Users] Re:Some SIP questions AGAIN gt;Hi Edwin
gt;I have my mgcp.conf almost the same as yours, except from nat=1 ,
why gt;do you put it? gt;Anyway, DL102s now works more or less
acceptably so now I'm having a gt;battle with sip.conf gt;Thank you
for your help gt;Michelle gt;- gt;Tu cuenta de correo gratuita Mixmail
A href=http://mixmail.ya.com/app/message?l=esamp;o=8amp;url=http%
3A%2F%2Fmixmail%2Eya%2Ecom target=_blankhttp://mixmail.ya.com/A
Ya.com ADSL gt;Home 24 h, Módem + Alta ¡Gratis! A
href=http://mixmail.ya.com/app/message?l=esamp;o=8amp;url=http%3A%
2F%2Facceso%2Eya%2Ecom%2Fadslhome24h%2F
target=_blankhttp://acceso.ya.com/adslhome24h//A
gt;___ gt;Asterisk-
Users mailing list gt;A href=javascript:sendMsg('Asterisk-
[EMAIL PROTECTED]');[EMAIL PROTECTED]/A gt;A
href=http://mixmail.ya.com/app/message?l=esamp;o=8amp;url=http%3A%
2F%2Flists%2Edigium%2Ecom%2Fmailman%2Flistinfo%2Fasterisk%
2Dusers target=_blankhttp://lists.digium.com/mailman/listinfo/asterisk-
users/A gt;___
gt;Asterisk-Users mailing list gt;A href=javascript:sendMsg('Asterisk-
[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-
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[Asterisk-Users] (no subject)
Hi 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)Whatis [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" 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" username=sip host=188.208.12.37 accountcode=sip Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] some sip questions
I write the email again, cause the first one I have had problems while sending it. Here is the email again: Hi 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" 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 = 5060bindaddr = 0.0.0.0context = defaulttransfer = yesthreewaycalling = yesusecallerid = yeshidecallerid = no [sip]type=friendcallerid="sip" username=siphost=188.208.12.37accountcode=sip Thanks you all!!! Michelle Tu cuenta de correo gratuita Mixmail Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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 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/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dl102s again
Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis - Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dl102S
I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
