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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