SIP Trunking HELP!!! 4

rhawkins74 · Feb 24, 2012

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.

NLPeterK · Feb 25, 2012

@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

rhawkins74 · Feb 25, 2012

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

NLPeterK · Feb 25, 2012

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.

NLPeterK · Feb 25, 2012

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

rhawkins74 · Feb 25, 2012

yes I see the call coming into the monitor.

rhawkins74 · Feb 25, 2012

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

NLPeterK · Feb 25, 2012

Set the PAI option to None.

I see you're calling as Anonymous.. I don't think they accept that.

rhawkins74 · Feb 25, 2012

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

NLPeterK · Feb 25, 2012

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

tlpeter · Feb 25, 2012

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!

rhawkins74 · Feb 25, 2012

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

tlpeter · Feb 25, 2012

Then you do not send the right information.

BAZINGA!

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

tlpeter · Feb 25, 2012

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!

rhawkins74 · Feb 25, 2012

so I would have to put in a uri for every user on a site?

NLPeterK · Feb 25, 2012

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.

rhawkins74 · Feb 25, 2012

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?

NLPeterK · Feb 25, 2012

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.

NLPeterK · Feb 25, 2012

Can you please let us know what did the trick?

rhawkins74 · Feb 25, 2012

what I did was in the SIP URI I changed the credentials to use credentials user name instead of use credentials auth name

NLPeterK · Feb 25, 2012

Never thought of that... in my systems those two fields are always the same ;-)

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SIP Trunking HELP!!! 4

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