Hello,
From my experience you could not make any improvements as this is an
Asterisk well known bug: in case you have multiple SIP accounts defined
to the same IP/hostname Asterisk will match the call to the first one it
will find in the configuration file (I think you will find more
information on this subject if you'll search the archive of the main
Asterisk list).
Nevertheless only the channel name is affected by this issue - the
routing it is OK.
As you could see:
Found peer '*7057974933*' for '6139929344' from *209.217.98.154:5060*
Executing [*7057974986*@default:2] Echo("*SIP/7057974933-0000000b*",
"") in new stack
Green = DID = OK
Red = Channel Name = Wrong info
In conclusion you should adapt your dialplan to overwrite some CDR
information (like accountcode) in order to be able to trace/bill
correctly the call to the right DID.
HTH,
Ioan
www.modulo.ro
On 27-May-14 4:23 AM, Shamus Rask wrote:
Lonnie,
Below is my Asterisk CLI output with “sip set debug on”; it is a call from my
mobile to 1 of the 2 DIDs I have registered with my ITSP. In my reading of the
output, it appears that even though I’ve called 705-797-4986, it is routed to
me via 705-797-4933.
Really scratching my head on this one… would really appreciate any pointers you
can give!
My SIP.conf includes:
[general]
context=default
register => 7057974933:[email protected]/7057974933
register => 7057974986:[email protected]/7057974986
[testtemplate](!)
type=peer
call-limit=5
disallow=all
allow=ulaw
qualify=yes
[7057974933](testtemplate) ; incoming calls
host=sip05.unlimitel.ca
fromuser=7057974933
remotesecret=PASSWORD1
canreinvite=no
[7057974986](testtemplate) ; incoming calls
host=sip05.unlimitel.ca
fromuser=7057974986
remotesecret=PASSWORD2
canreinvite=no
<--- SIP read from UDP:209.217.98.154:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK57d6f0e1;rport
Max-Forwards: 70
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Primus-Unlimitel
Date: Tue, 27 May 2014 00:37:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 1448758994 1448758994 IN IP4 209.217.98.154
s=Primus-Unlimitel
c=IN IP4 209.217.98.154
t=0 0
m=audio 17652 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 209.217.98.154:5060 (no NAT)
Sending to 209.217.98.154:5060 (no NAT)
Using INVITE request as basis request -
[email protected]
Found peer '7057974933' for '6139929344' from 209.217.98.154:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer -
audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 209.217.98.154:17652
Looking for 7057974986 in default (domain 198.72.123.133)
list_route: hop: <sip:[email protected]>
<--- Transmitting (no NAT) to 209.217.98.154:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.217.98.154:5060;branch=z9hG4bK57d6f0e1;received=209.217.98.154;rport=5060
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [7057974986@default:1] Answer("SIP/7057974933-0000000b", "")
in new stack
Audio is at 25012
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 209.217.98.154:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.217.98.154:5060;branch=z9hG4bK57d6f0e1;received=209.217.98.154;rport=5060
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>;tag=as4a7817dc
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253
v=0
o=root 1177512703 1177512703 IN IP4 198.72.123.133
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 198.72.123.133
t=0 0
m=audio 25012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:209.217.98.154:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK4b7802d3;rport
Max-Forwards: 70
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>;tag=as4a7817dc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Primus-Unlimitel
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Executing [7057974986@default:2] Echo("SIP/7057974933-0000000b", "") in
new stack
<--- SIP read from UDP:209.217.98.154:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK5d991769;rport
Max-Forwards: 70
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>;tag=as4a7817dc
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Primus-Unlimitel
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 209.217.98.154:5060 (no NAT)
Scheduling destruction of SIP dialog
'[email protected]' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 209.217.98.154:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.217.98.154:5060;branch=z9hG4bK5d991769;received=209.217.98.154;rport=5060
From: "6139929344" <sip:[email protected]>;tag=as65cd906e
To: <sip:[email protected]:5060>;tag=as4a7817dc
Call-ID: [email protected]
CSeq: 103 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (default, 7057974986, 2) exited non-zero on
'SIP/7057974933-0000000b'
Shamus,
'type=peer' is what you want for each DID's sip context. You probably need to
register for each DID as well for your situation.
Lonnie
On 2014 May 26, at 10:50, Shamus Rask <[email protected]> wrote:
Lonnie,
Yes, I should have specified that each DID has it’s own user/password (user is
just the DID number). Should I be looking at a type= other than peer? Am I
looking down the right track?
many thanks,
Shamus
Message: 5
Date: Mon, 26 May 2014 09:27:57 -0500
From: Lonnie Abelbeck <[email protected]>
Subject: Re: [Astlinux-users] help with multiple DID/SIP trunks
To: AstLinux Users Mailing List <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=windows-1252
Hi Shamus,
Does the unlimitel.ca service allow sub-accounts for DID's, such that unique
user/pass credentials could be specified for each DID. This requires
additional configuration in sip.conf but should get you around the aggregated 5
channel limit.
You might also inquire why the 5 channel limit exists, sometimes that is a
default sanity setting by the SIP provider and can be raised at no additional
cost.
Lonnie
On May 26, 2014, at 8:30 AM, Shamus Rask wrote:
I?m posting here as I find the AstLinux community to be the most friendly and
knowledgeable about all things Asterisk!
My ITSP offers DID/SIP trunks at a very competitive rate?each DID includes 5
channels. I?m running Asterisk 11.
The ITSP only offers a single server for both incoming and outgoing calls:
sip05.unlimitel.ca. I recently discovered that with the configuration I had, all of
my calls, no matter how many DIDs I have, were being sent over a single trunk (the
first to register from Asterisk). I believe this is due to my selecting type=peer
in my SIP.conf; it appears to match based on IP & port so all of the DID/trunks
appear as a single one. This means that I?m limited to 5 channels, despite having
4xDIDs which should give me 20 channels (5 per DID).
I?ve tried changing this to type=user, but although I see registration (sip
show registry) and users (sip show users) I cannot see any incoming calls.
Can anyone offer any help/suggestions? Pulling my hair out!
Shamus
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