Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
Oh wow, look at that. Don't I feel stupid, haha. All right, it works now, thanks! :/ Matt - Original Message - From: Robert Webb To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, February 05, 2005 12:05 PM Subject: RE: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone Matt, I thought that DIAX was an IAX based phone not SIP based. If this is the case then you need to be putting your configs in the iax.conf not sip.conf file. I have several iax soft phones I have been testing and have them registering with asterisk. If you want, I can email you the config I have for them off-list. Robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt WatermanSent: Saturday, February 05, 2005 11:38 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone Thanks for the encouraging advice. I actually spent many hours searching for and reading through documentation about this (on the wiki and in the handbook) and I couldn't figure out how Asterisk was supposed to work as an SIP server. Since I posted my original message I've made a lot more progress (and spent considerably more than 15 minutes) but I still have not managed to get it to work. I have specified an SIP extension (many, actually) in the sip.conf file but I cannot get DIAX to register with Asterisk. I've tried changing just about every variable I can while troubleshooting. One thing that is kind of suspect is what comes up after I have it re-read the config files: -- Messages-Waiting: no Voice-Message: 0/0 to 192.168.9.102:5060 Retransmitting #5 (no NAT): NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK7cc5dc1e From: "Unknown" ;tag=as63d4a421 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message=summary Content-Type: application/simple-message-summary Content-Length: 42 -- 192.168.9.101 is the Asterisk server and 192.168.9.102 is the machine I've been trying to get DIAX registered on. In the past, I've specified the .102 address in the SIP config file for an extension but at this point I can't think of anywhere where that IP address is specified so this is a big mystery to me. Can anyone make sense of it? I have the following users in my sip_additionals file (as generated by AMP): [200]username=200type=friendsecret=testqualify=noport=5060nat=never mailbox=200host=dynamicdtmfmode=infocontext=from-internalcanreinvite=nocallerid="test" <200> [222]username=222 type=friendsecret=222qualify=noport=5060nat=nevermailbox=556host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="Jack" <222> And I've tried making simpler a simpler one with the bare minimum: [111] username=111 type=friend secret=111 port=5060 I haven't been able to register with any of these. I'm probably missing something really simple, I'm sure, but I haven't been able to find it in all of the time I've spent and I imagine it would take someone less time to point it out to me than it would to write a message telling me how I shouldn't have posted. Matt - Original Message - From: Chamberland-Larose, Guillaume To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, February 03, 2005 2:28 PM Subject: RE: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone I believe the web page should be modified to include a huge, red, bold, blinking "please read the asterisk handbook available here and search the wiki and mail archives before you post a message to the list". That would prevent so many questions on how and where to start when first installing asterisk. :s So, I would suggest you check out the asterisk handbook here: http://www.digium.com/handbook-draft.pdf Page 56 to 61 explain in lots of detail and give a working example of sip.conf with 1 phone and 1 voip provider. The whole thing is good to read though so you might as well read the whole thing (quickly) hehe. The handbook assumes you know nothing about asterisk and pretty much everything else. You shouldn't have to spend more than 15 minutes configuring this. Guills From: Matt Waterman [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 02, 2005 7:08 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Calling Asterisk Autoattendant With SIP
Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
> I have specified an SIP extension (many, actually) in the sip.conf file but > I cannot get DIAX to register with Asterisk. I've tried changing just about > every variable I can while troubleshooting. One thing that is kind of > suspect is what comes up after I have it re-read the config files: you are configuring it in the wrong place. DIAX is not an SIP phone, it's a IAX phone setup your account in iax.conf or, get a sip phone, like X-lite : http://www.xten.com/index.php?menu=products&smenu=xlite hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
Matt,
I thought that DIAX was an IAX based phone not
SIP based. If this is the case then you need to be putting your configs in the
iax.conf not sip.conf file. I have several iax soft phones I have been testing
and have them registering with asterisk. If you want, I can email you the config
I have for them off-list.
Robert
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
WatermanSent: Saturday, February 05, 2005 11:38 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone
Thanks for the encouraging advice. I actually spent
many hours searching for and reading through documentation about this (on the
wiki and in the handbook) and I couldn't figure out how Asterisk was supposed to
work as an SIP server.
Since I posted my original message I've made a lot
more progress (and spent considerably more than 15 minutes) but I still have not
managed to get it to work.
I have specified an SIP extension (many, actually)
in the sip.conf file but I cannot get DIAX to register with Asterisk. I've tried
changing just about every variable I can while troubleshooting. One thing that
is kind of suspect is what comes up after I have it re-read the config
files:
--
Messages-Waiting: no
Voice-Message: 0/0
to 192.168.9.102:5060
Retransmitting #5 (no NAT):
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.9.101:5060;branch=z9hG4bK7cc5dc1e
From: "Unknown"
;tag=as63d4a421
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message=summary
Content-Type:
application/simple-message-summary
Content-Length: 42
--
192.168.9.101 is the Asterisk server and
192.168.9.102 is the machine I've been trying to get DIAX registered on. In the
past, I've specified the .102 address in the SIP config file for an extension
but at this point I can't think of anywhere where that IP address is specified
so this is a big mystery to me. Can anyone make sense of it?
I have the following users in my sip_additionals
file (as generated by AMP):
[200]username=200type=friendsecret=testqualify=noport=5060nat=never
mailbox=200host=dynamicdtmfmode=infocontext=from-internalcanreinvite=nocallerid="test"
<200>
[222]username=222
type=friendsecret=222qualify=noport=5060nat=nevermailbox=556host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="Jack"
<222>
And I've tried making simpler a simpler one with
the bare minimum:
[111]
username=111
type=friend
secret=111
port=5060
I haven't been able to register with any of these.
I'm probably missing something really simple, I'm sure, but I haven't been able
to find it in all of the time I've spent and I imagine it would take someone
less time to point it out to me than it would to write a message telling me how
I shouldn't have posted.
Matt
- Original Message -
From:
Chamberland-Larose,
Guillaume
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 03, 2005 2:28
PM
Subject: RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here
and search the wiki and mail archives before you post a message to the list".
That would prevent so many questions on how and where to start when first
installing asterisk. :s
So, I would suggest you check out the asterisk handbook
here: http://www.digium.com/handbook-draft.pdf
Page 56 to 61 explain in lots of detail and give a
working example of sip.conf with 1 phone and 1 voip provider. The whole thing
is good to read though so you might as well read the whole thing (quickly)
hehe.
The handbook assumes you know nothing about asterisk and
pretty much everything else. You shouldn't have to spend more than 15 minutes
configuring this.
Guills
From: Matt Waterman
[mailto:[EMAIL PROTECTED] Sent: Wednesday, February 02, 2005
7:08 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone
I'm trying to get into the world of
Asterisk in order to use the voicemail and autoattendat features (and more
stuff later) with a Redcom switch. But, I've only started and haven't gotten
to that yet. At this point my solitary goal is to talk to the autoattendant
via an SIP phone (SJPhone). I've spent countless hours trying to find the
documentation I need to accomplish my goals but everything I find always
assumes so much and I'm left lost. Plus I haven't found a thing about
setting up Asterisk as an SIP server.
I installed the [EMAIL PROTECTED] package, so I can edit all the
Re: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
Thanks for the encouraging advice. I actually spent
many hours searching for and reading through documentation about this (on the
wiki and in the handbook) and I couldn't figure out how Asterisk was supposed to
work as an SIP server.
Since I posted my original message I've made a lot
more progress (and spent considerably more than 15 minutes) but I still have not
managed to get it to work.
I have specified an SIP extension (many, actually)
in the sip.conf file but I cannot get DIAX to register with Asterisk. I've tried
changing just about every variable I can while troubleshooting. One thing that
is kind of suspect is what comes up after I have it re-read the config
files:
--
Messages-Waiting: no
Voice-Message: 0/0
to 192.168.9.102:5060
Retransmitting #5 (no NAT):
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.9.101:5060;branch=z9hG4bK7cc5dc1e
From: "Unknown"
;tag=as63d4a421
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message=summary
Content-Type:
application/simple-message-summary
Content-Length: 42
--
192.168.9.101 is the Asterisk server and
192.168.9.102 is the machine I've been trying to get DIAX registered on. In the
past, I've specified the .102 address in the SIP config file for an extension
but at this point I can't think of anywhere where that IP address is specified
so this is a big mystery to me. Can anyone make sense of it?
I have the following users in my sip_additionals
file (as generated by AMP):
[200]username=200type=friendsecret=testqualify=noport=5060nat=never
mailbox=200host=dynamicdtmfmode=infocontext=from-internalcanreinvite=nocallerid="test"
<200>
[222]username=222
type=friendsecret=222qualify=noport=5060nat=nevermailbox=556host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="Jack"
<222>
And I've tried making simpler a simpler one with
the bare minimum:
[111]
username=111
type=friend
secret=111
port=5060
I haven't been able to register with any of these.
I'm probably missing something really simple, I'm sure, but I haven't been able
to find it in all of the time I've spent and I imagine it would take someone
less time to point it out to me than it would to write a message telling me how
I shouldn't have posted.
Matt
- Original Message -
From:
Chamberland-Larose,
Guillaume
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 03, 2005 2:28
PM
Subject: RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here
and search the wiki and mail archives before you post a message to the list".
That would prevent so many questions on how and where to start when first
installing asterisk. :s
So, I would suggest you check out the asterisk handbook
here: http://www.digium.com/handbook-draft.pdf
Page 56 to 61 explain in lots of detail and give a
working example of sip.conf with 1 phone and 1 voip provider. The whole thing
is good to read though so you might as well read the whole thing (quickly)
hehe.
The handbook assumes you know nothing about asterisk and
pretty much everything else. You shouldn't have to spend more than 15 minutes
configuring this.
Guills
From: Matt Waterman
[mailto:[EMAIL PROTECTED] Sent: Wednesday, February 02, 2005
7:08 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone
I'm trying to get into the world of
Asterisk in order to use the voicemail and autoattendat features (and more
stuff later) with a Redcom switch. But, I've only started and haven't gotten
to that yet. At this point my solitary goal is to talk to the autoattendant
via an SIP phone (SJPhone). I've spent countless hours trying to find the
documentation I need to accomplish my goals but everything I find always
assumes so much and I'm left lost. Plus I haven't found a thing about
setting up Asterisk as an SIP server.
I installed the [EMAIL PROTECTED] package, so I can edit all the
config files through HTTP and I can use AMP.
I've tried 'dialing' to the IP address of
the Asterisk machine with SJPhone but the call is rejected ("number not
available"). Now, how do I specify an extension number when I
'dial'?
Thanks for any help :/
Matt
___Asterisk-Users
mailing
listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
U
RE: [Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here and
search the wiki and mail archives before you post a message to the list". That
would prevent so many questions on how and where to start when first installing
asterisk. :s
So, I would suggest you check out the asterisk handbook
here: http://www.digium.com/handbook-draft.pdf
Page 56 to 61 explain in lots of detail and give a working
example of sip.conf with 1 phone and 1 voip provider. The whole thing is good to
read though so you might as well read the whole thing (quickly)
hehe.
The handbook assumes you know nothing about asterisk and
pretty much everything else. You shouldn't have to spend more than 15 minutes
configuring this.
Guills
From: Matt Waterman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 7:08 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
I'm trying to get into the world of Asterisk
in order to use the voicemail and autoattendat features (and more stuff later)
with a Redcom switch. But, I've only started and haven't gotten to that yet.
At this point my solitary goal is to talk to the autoattendant via an SIP
phone (SJPhone). I've spent countless hours trying to find the documentation I
need to accomplish my goals but everything I find always assumes so much and
I'm left lost. Plus I haven't found a thing about setting up Asterisk as an
SIP server.
I installed the [EMAIL PROTECTED] package, so I can edit all the
config files through HTTP and I can use AMP.
I've tried 'dialing' to the IP address of
the Asterisk machine with SJPhone but the call is rejected ("number not
available"). Now, how do I specify an extension number when I
'dial'?
Thanks for any help :/
Matt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling Asterisk Autoattendant With SIP Phone
I'm trying to get into the world of Asterisk
in order to use the voicemail and autoattendat features (and more stuff later)
with a Redcom switch. But, I've only started and haven't gotten to that yet. At
this point my solitary goal is to talk to the autoattendant via an SIP phone
(SJPhone). I've spent countless hours trying to find the documentation I need to
accomplish my goals but everything I find always assumes so much and I'm left
lost. Plus I haven't found a thing about setting up Asterisk as an SIP
server.
I installed the [EMAIL PROTECTED] package, so I can edit all the
config files through HTTP and I can use AMP.
I've tried 'dialing' to the IP address of the
Asterisk machine with SJPhone but the call is rejected ("number not available").
Now, how do I specify an extension number when I 'dial'?
Thanks for any help :/
Matt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
