Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi BK, Using your configuration info, I now have Nikotel working again. Other than the fromuser=, it appears that one also now needs the auth=md5 whereas before it was not necessary. To disable incoming calling, just delete the register -> line for Nikotel. That way, no one can find you. You do not need to the register -> line for outgoing calls. On Monday, July 7, 2003, at 11:16 AM, BK [address only for mailing lists] wrote: Hi Paul, thanks for your insights On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote: To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!). This is what we have discovered last night. However, We have got it working now. I will document this in detail and make it available, but briefly here a quick summary ... First I had various glitches in my dial string. With the help of John Todd and some others on the IRC #asterisk channel I was able to fix those glitches. Thanks everybody who assisted. Then I tried a number of things I had already experimented with before. When I turned on SIP debug and watched the datagrams, I could see Nikotel's response "account name does not match address of record". Together with the "from" part, this led me to fiddle with "fromuser" again and when I set it to the actual login name, it > worked. Their tech people said it should work with a slight change: "yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default." This CAN be done if you edit chan_sip.c, It would seem you can do it a lot simpler: in sip.conf --- -- register => myusername:[EMAIL PROTECTED] [nikotel] username=myusername fromuser=myusername ... --- -- but when I did this, it billed me a few times for unconnected calls Thanks for sharing this with us. I will watch this for a while and see if this happens here too. and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to. I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded. Yes, support is not exactly their strength, is it?! FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. I don't want anybody to call in via Nikotel. Since they do not provide a telephone number for incoming calls, the only calls you could possibly get are from their public chat room. In the very best case you get a friendly test call from somebody who has just signed up and wants to try out the service, in the worst case you get prank calls in the middle of the night or indecent proposals and all the rest of it. I will have to find a way to disable incoming calls from Nikotel entirely. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Yes, I have seen that. And at $8.95/mth it would seem reasonably priced, too. Outbound calls do not require registering. I can provide examples of iConnect connection scripts if you contact me offline. Thanks, I will do that. again many thanks to everybody who has helped solving this riddle rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi Paul, thanks for your insights On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote: To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!). This is what we have discovered last night. However, We have got it working now. I will document this in detail and make it available, but briefly here a quick summary ... First I had various glitches in my dial string. With the help of John Todd and some others on the IRC #asterisk channel I was able to fix those glitches. Thanks everybody who assisted. Then I tried a number of things I had already experimented with before. When I turned on SIP debug and watched the datagrams, I could see Nikotel's response "account name does not match address of record". Together with the "from" part, this led me to fiddle with "fromuser" again and when I set it to the actual login name, it worked. Their tech people said it should work with a slight change: "yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default." This CAN be done if you edit chan_sip.c, It would seem you can do it a lot simpler: in sip.conf - register => myusername:[EMAIL PROTECTED] [nikotel] username=myusername fromuser=myusername ... - but when I did this, it billed me a few times for unconnected calls Thanks for sharing this with us. I will watch this for a while and see if this happens here too. and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to. I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded. Yes, support is not exactly their strength, is it?! FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. I don't want anybody to call in via Nikotel. Since they do not provide a telephone number for incoming calls, the only calls you could possibly get are from their public chat room. In the very best case you get a friendly test call from somebody who has just signed up and wants to try out the service, in the worst case you get prank calls in the middle of the night or indecent proposals and all the rest of it. I will have to find a way to disable incoming calls from Nikotel entirely. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Yes, I have seen that. And at $8.95/mth it would seem reasonably priced, too. Outbound calls do not require registering. I can provide examples of iConnect connection scripts if you contact me offline. Thanks, I will do that. again many thanks to everybody who has helped solving this riddle rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi,
To dial a PSTN number through Nikotel used to work from Asterisk, but
they had a very serious security issue (you could make calls anytime
anywhere and their billing wouldn't charge it) and after I informed
them of this, they changed their authentication mechanism and since
then I have not gotten it to work (they didn't even thank me!).
Their tech people said it should work with a slight change: "yes, we
changed it yesterday. Now the user part of the From: address has to be
the same as the username in the Proxy-Authentication line. I don't know
if the Asterisk can do that. The ATA186 does it b[y] default."
This CAN be done if you edit chan_sip.c, but when I did this, it billed
me a few times for unconnected calls and I gave up trying to debug and
switched to iConnect. iConnect is worse quality, but it is very easy to
connect to.
I had much better quality with calls via Nikotel than iConnect, but
their support is non-existent/bad at best. I sent them 3-4 e-mails
about their security issue before they even responded.
FYI. Registering with Nikotel was futile anyways, because I never
figured out how anyone could call into me. iConnect provides a PSTN-SIP
dial in as an option, but I haven't tried it. Outbound calls do not
require registering.
I can provide examples of iConnect connection scripts if you contact me
offline.
On Saturday, July 5, 2003, at 07:42 PM, BK [address only for mailing
lists] wrote:
Hi
thanks to everybody who responded to my earlier post. I have looked at
all the material and links provided and tried everything in there, but
it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call anything (error
code 4, whatever that means) not even internal (yes I did give them
appropriate context).
Further, Asterisk registers with my VoIP provider via SIP just fine,
but I cannot make any calls even from the analog phones.
sip show registry gives me
HostUsernameRefresh State
63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
I wonder what the syntax is to dial a number via a VoIP provider. This
appears to be documented NOWHERE.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on
record, whatever that means.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
exten => _00N.,2,Congestion
and this doesn't even show anything but immediately gives me a busy
signal. The fact that there is no debugging output leads me to believe
that Asterisk didn't even attempt to try talking to the VoIP server.
Does anybody know how to dial a PSTN number through a VoIP service?
Is this standardised, at least within SIP? Or does it vary from
provider to provider?
any hints appreciated
kind regards
bk
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---
Paul Cheng
Mátyás király ut 10
H-1121 Budapest HUNGARY
[EMAIL PROTECTED]
mobile: +36 30 381-9311
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Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi
thanks for your reply.
On Sunday, July 6, 2003, at 03:44 AM, John Todd wrote:
sip show registry gives me
HostUsernameRefresh State
63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
The command you will find more useful is "sip show peers".
show peers returns
Name/username HostMaskPort Status
Sip1/gsone 192.168.0.160 255.255.255.255 5060 Unmonitored
Sip2/gstwo 192.168.0.161 255.255.255.255 5060 Unmonitored
nikotel/myusername 63.214.186.6255.255.255.255 5060 Unmonitored
I don't think the problem is with registration. As I said, sip debug
confirms successful registration:
Sip read:
SIP/2.0 200 OK - 'Authenticated'
Via: SIP/2.0/UDP 192.168.0.111;branch=weoiuewotu
From: ;tag=xznxzcvnxzvn
To:
CSeq: 121 REGISTER
Call-ID: [EMAIL PROTECTED]
Expires: 120
Content-Length: 0
Contact:
If your hosts are "(Unspecified)" then your SIP clients are not
registering, and inbound calls will not work if you are using
"dynamic=yes" in your sip.conf. Possibly it may be helpful if you
would statically register your SIP phones until you get things working
better ("host=10.3.2.3" in sip.conf)
I had already done that. Part of the problem seems to have been the
length of the identifiers I had defined.
[Grandstream1]
username=grandstream1
appears to have been too long, so I shortened it to
[Sip1]
username=gsone
Now I can use the SIP phones internally at least. I can call from Sip1
to Zap2 for example. Dialing into the PSTN from the Sip phones is still
problematic, ie calls dropping, tiny volume etc etc.
However, for as long as I cannot even dial into Nikotel (nor FWD nor
ICH) using one of the analog phones on (Zap2 or Zap3) I don't even fancy
to fiddle with the SIP phones. The fever things there are in the call
chain that could be responsible for it not working, the better, so for
now, I want to concentrate on getting the dialing out via VoIP service
provider working from the Zap lines.
I wonder what the syntax is to dial a number via a VoIP provider. This
appears to be documented NOWHERE.
I would disagree. A VoIP provider is no different than a SIP phone;
they are treated the same.
If I dial into a SIP phone directly, I don't have to provide a third
party number because the SIP phone is the destination already. If I dial
into a VoIP service, I have to provide the number for my desired
destination in addition to my own credentials. As a result the two cases
are not the same as far as dialing syntax is concerned.
If you are looking for examples, please see
http://www.loligo.com/asterisk/ for my sample files, which contain
some VoIP provider dial statements.
Thanks, I had already been given the URL and I looked at it. One of the
problems I had is that I find it difficult to work out what the *naked*
dial string actually is because of all the macros and variables used in
there.
I would prefer to start as barebones as it can possibly be and get the
basics working. Bells and whistles can be added in later - one at a time.
exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on
record, whatever that means.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
You may be having at least one error due to syntax. The line above
should look like:
exten =>
_00N.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:2},100,r)
Thanks for the hint. However, I am not quite so sure that this is the
correct syntax either. When I try this with "sip debug" Asterisk does
not even make an attempt to contact the remote VoIP service. It chokes
on the dial string already and goes no further.
SIP Registration comes back with a SIP/2.0 200 OK (ie registration
successful) for both ICH and Nikotel. Thus, I am confident that
registration is not the issue. However dialing out (from Zap) with the
various possibilities of syntax I could think of, simply doesn't work.
Here is what I have tried so far
in zapata.conf ...
-
signalling=fxo_ks
context=internal
channel => 2,3
-
in sip.conf ...
-
register => user:[EMAIL PROTECTED]/asterisk ; already tried with
user instead of asterisk and blank
[nikotel] ; Service Provider Nikotel
type=peer
secret=pass
username=user
host=calamar0.nikotel.com
-
in extensions.conf ...
-
[globals]
REDPHONE => Zap/3
[voipintl]
;
; International long distance through VoIP ser
[Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi
thanks to everybody who responded to my earlier post. I have looked at
all the material and links provided and tried everything in there, but
it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call anything (error code
4, whatever that means) not even internal (yes I did give them
appropriate context).
Further, Asterisk registers with my VoIP provider via SIP just fine, but
I cannot make any calls even from the analog phones.
sip show registry gives me
HostUsernameRefresh State
63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
I wonder what the syntax is to dial a number via a VoIP provider. This
appears to be documented NOWHERE.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on record,
whatever that means.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
exten => _00N.,2,Congestion
and this doesn't even show anything but immediately gives me a busy
signal. The fact that there is no debugging output leads me to believe
that Asterisk didn't even attempt to try talking to the VoIP server.
Does anybody know how to dial a PSTN number through a VoIP service?
Is this standardised, at least within SIP? Or does it vary from provider
to provider?
any hints appreciated
kind regards
bk
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