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SIP Trunking HELP!!! 4

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rhawkins74

rhawkins74

Vendor
Joined
Nov 5, 2010
Messages
64
Location
US
Hey everyone...
Well here is my issue, I am in the process of setting up my first SIP trunks. This is a completely new area for me, and have to be honest I am lost. I definitely need some help on this, I don't even know where to start.
 
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@amriddle01: REALLY!? OMG! It's been my frustration since I started doing IPO on R4. Never understood why SIP had to be completely different from ISDN/analogue regarding ARS codes etc...

And now I completely missed that they sort of cleaned it up :P
 
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Channel 1
Via 10.0.1.190
Local URI: Use Credentials User Name
Contact:Use Credentials User Name
Display Name: Use Credentials User Name
PAI: Use Credentials User Name
Registration set to SIP Credential
Incoming Group: 0
Outgoing Group: 2
Max Calls per Channel: 2

ARS
Route Id 51
Route Name SIP
Dial Delay Time System default
N;/N/Dial/2

Shortcode
Code: 8N
Feature: Dial
Telephone Number: N
LineGroup ID: 51:SIP

 
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In theory this looks good, but looking back at your previous INVITE messages it seems the IPO is sending the actual letter N instead of the number you typed...

Curious.
 
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Do you actually see a call coming in in the monitor?

I think this has to do with the Use Client Authentication Name... that's why advised the separate URI with stars :-)
 
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yes I see the call coming into the monitor.
 
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This is what I get in the monitor when trying to place a call:


665807mS Sip: License, Valid 1, Available 10, Consumed 0
665861mS Sip: CMMediaSTUNFilter::callback_received addr f5a65494 (rtp f5a65494 rtcp f5a65380) rec rtp 0 rtcp 0 video rtp 1 video rtcp 1
667859mS Sip: CMMediaSTUNFilter::callback_received addr f5a65380 (rtp f5a65494 rtcp f5a65380) rec rtp 1 rtcp 0 video rtp 1 video rtcp 1
667861mS Sip: CMMediaSTUNFilter substituting
667861mS Sip: 17.1018.0 4 SIPTrunk Endpoint(f52a86a0) received CMSetup
667862mS Sip: 17.1018.0 4 SIPTrunk Endpoint(f52a7484) SetLocalRTPAddress to 97.64.214.34:49152 (index 0)
667864mS SIP Call Tx: 17
INVITE sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Asserted-Identity: "advtt44534476455" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Content-Length: 204

v=0
o=UserA 4138983088 1166865868 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
667864mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
INVITE sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Asserted-Identity: "advtt44534476455" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Content-Length: 204

v=0
o=UserA 4138983088 1166865868 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
667935mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE

667937mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE

667939mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-fqshamodsnvoa
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE

667940mS SIP Call Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-fqshamodsnvoa
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 INVITE

667942mS SIP Call Tx: 17
ACK sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-fqshamodsnvoa
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 ACK
Privacy: id
P-Asserted-Identity: "advtt44534476455" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0

667943mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
ACK sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bKb572f5d3f263b6cf6a7ece98a46f2159
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=78d163eb166d0438
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-fqshamodsnvoa
Call-ID: e85cec03819d6d2897d5a804eb3714d8@97.64.214.34
CSeq: 1862041260 ACK
Privacy: id
P-Asserted-Identity: "advtt44534476455" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0
 
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Set the PAI option to None.

I see you're calling as Anonymous.. I don't think they accept that.
 
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okay, I set PAI to None and still recieving the same thing


134568mS Sip: License, Valid 1, Available 10, Consumed 0
134622mS Sip: CMMediaSTUNFilter::callback_received addr f5a65380 (rtp f5a65494 rtcp f5a65380) rec rtp 0 rtcp 0 video rtp 1 video rtcp 1
142530mS PRN: -----------Collecting System Information--------------
142530mS PRN: Platform::CollectSystemInformationOutput No Date
166625mS Sip: CMMediaSTUNFilter::callback_received addr f5a65494 (rtp f5a65494 rtcp f5a65380) rec rtp 0 rtcp 1 video rtp 1 video rtcp 1
166626mS Sip: CMMediaSTUNFilter substituting
166626mS Sip: 17.1010.0 2 SIPTrunk Endpoint(f52a31e0) received CMSetup
166627mS Sip: 17.1010.0 2 SIPTrunk Endpoint(f52a1fc4) SetLocalRTPAddress to 97.64.214.34:49152 (index 0)
166629mS SIP Call Tx: 17
INVITE sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Preferred-Identity: "Unavailable" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Content-Length: 203

v=0
o=UserA 530056570 1387573597 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
166629mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
INVITE sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Preferred-Identity: "Unavailable" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Content-Length: 203

v=0
o=UserA 530056570 1387573597 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
166699mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE

166701mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE

166702mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-1of9tvm514lde
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE

166704mS SIP Call Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-1of9tvm514lde
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 INVITE

166706mS SIP Call Tx: 17
ACK sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-1of9tvm514lde
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 ACK
Privacy: id
P-Preferred-Identity: "Unavailable" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0

166707mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
ACK sip:12178557935@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK787cd8a4407bd2bab8639b85bd5d470a
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9c90ccfd9dcbf41e
To: <sip:12178557935@sbc.voipdnsservers.com>;tag=aprqngfrt-1of9tvm514lde
Call-ID: 9b38e3ce01ac072f16426bb1dfc31977@97.64.214.34
CSeq: 763483366 ACK
Privacy: id
P-Preferred-Identity: "Unavailable" <sip:advtt44534476455@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0
 
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Guess I really need to see your full config.

Can you post or mail your .cfg? I'll have a look later! Get some sleep :-)
 
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I keep seeing a 403 forbidden.
Are you sure you have entered the right credentials?

BAZINGA!

I'm not insane, my mother had me tested!

 
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tlpeter.. well, since it is registering, I would assume it is. I am now able to receive calls on the sip trunks, just not make them
 
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Then you do not send the right information.


BAZINGA!

I'm not insane, my mother had me tested!

 
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Go to a user and see if it has a tab for sip.
When it does then put in a sip uri and again with that user.


BAZINGA!

I'm not insane, my mother had me tested!

 
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so I would have to put in a uri for every user on a site?
 
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No, although that is what the older documentation suggests.

if you like at my length post:

"You can now make as many URI's as you like for outgoing calls, all with different CLI's. If you put it on Use internal data, it sends the number you specifiy in the user. I find this handy if I need to send personal DID's. Make sure you specify a different Outgoing Line ID."

The part Use Internal Data allows you to send CLI per user. It will then send the data stored under the SIP tab in the user.


What did you put under that tab where you store username and password? It has three fields and a password field. I usually fill all three fields with the username.

Since you're using "Use Client Authentication Name", the call setup will take all its info from there.

If you use "Use Internal Data" it will take it from the user.
 
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okay, I have now gotten it to make and recieve calls. Since I have 2 trunks should I set the max calls at 2 for the SIP URI or should I create a second channel?
 
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You probably mean that you have a sip trunk with 2 channels? :)

I personally never limit the IPO, because the provider already has that limit in place.

If you put a limit in the IPO and you decide you need to expand it would require a reboot to up the channels.

If the client has a trunk with 10 channels, I usually put the IPO limit on 15-20 per URI.
 
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Can you please let us know what did the trick? ;)
 
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what I did was in the SIP URI I changed the credentials to use credentials user name instead of use credentials auth name
 
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Never thought of that... in my systems those two fields are always the same ;-)
 
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