SPA3102 VOIP to PSTN not working

[a123a123a123]

Hello!

I'm using a Fritz-box 7390 as DECT base. Because my austrian telecomunications provider blocks the usage of the Fritz-Modem's phone connector I can only use this device via VOIP:

In order to use my physical landline I bought a SPA 3102 connected to the second

telecomunications provider's FXS where before a plain old telephone was working.

My plan to dial out using physical land line is the following - please confirm if this is possible:

a) Dial a number at my DECT phone ==> Fritz-box 7390 ==> VOIP account 1234@somesipprovider.at

OR

b) Dial a number at Couterpath's X-Lite using my PC ==> VOIP account 1234@somesipprovider.at

==>

1234@somesipprovider.at ==> VOIP Line1 at SPA 3102 ==> PSTN at  SPA 3102 ==> SPA 3102 dials out using physical land line

I configured the SPA 3102 using VOIP account 1234@somesipprovider.at according to the following tutorial: Making an outbound call on SPA3102

http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a359e7.shtml

Later I confgured a VOIP Number using account 1234@somesipprovider.at at the Fritz box or on PC with X-Lite from Couterpath.

However everytime I dial using the VOIP provider I phone out directly with the provider instead of invoking  the VOIP to PSTN Gateway  from SPA3102

How is the configuration on both sides, the VOIP-sender e.g using X-Lite or Fritz-Box and the VOIP-receiver (SPA3102)

Thank you for help and your reply!

KR H.

[Howard Wittenberg]

The tutorial you followed looks to me like it was for dialing a voip call and for dialing #9 to dial a PSTN Line call both from the phone attached to the SPA3102.  That is not what you are trying to do.

You are trying to call the SPA3102 and bridge that call with an outgoing call on the PSTN line.

You setup the SPA3102 PSTN Line Tab (not the Line 1 Tab) with a voip account and setup the account to Register.

NAT Mapping Enable: Yes/No (depending on behind a router)

Proxy: your_sip_proxy

Register: Yes

User ID: your_userid

Password: your_password

Setup the VoIP-To-PSTN Gateway to return a dial tone when you call the voip account:

VoIP-To-PSTN Gateway Enable: yes

VoIP Caller Auth Method: none

One Stage Dialing: yes

VoIP Caller Default DP: 1

Dial Plan 1: (xx.)

Note:  A PSTN line must be attached to the SPA3102 Line jack.

[a123a123a123]

Hello 99hwittenb!

Thank you for  your reply!

I'm intending to use the SPA3102 with 2 functionalities:

a) PSTN ==> VOIP: works, however caller ID not shown (see https://supportforums.cisco.com/message/3871352#3871352)

b) VOIP ==> PSTN (this posting, currently not working)

Thank you for the hints, which I reviewed:

> SPA3102 PSTN Line Tab

> NAT Mapping Enable: Yes ( I'm behind the Fritzbox ADSL Modem, however I use the SPA3102's internal bridge functionality for the ethernet port)

> Proxy: your_sip_proxy  ok

> Register: Yes  ok

> User ID: your_userid  ok

> Password: your_password ok

>

> Setup the VoIP-To-PSTN Gateway to return a dial tone when you call the voip account:

>

> VoIP-To-PSTN Gateway Enable: yes

> VoIP Caller Auth Method: none

> One Stage Dialing: yes

> VoIP Caller Default DP: 1

>

> Dial Plan 1: (xx.)

>

> A PSTN line must be attached to the SPA3102 Line jack: is connected

See also below config of SPA3102's PSTN Line tab

Behaviour didn't change:

a) when I call a number (by directly tpying in) with the DECT phone: call is using SIP account to phone directly

b) when I call a number

(by directly tpying in) with X-Lite: same here, call is using SIP account to phone directly

How do I configure e.g X-Lite to bridge it to my SPA3102 for calling out? Is there a special configuration necessary?

Thanks for your reply!

KR H.

Router Voice PSTN Line Configuration settings:

Line Enable: yes

NAT Settings

NAT Mapping Enable: yes NAT Keep Alive Enable: no

NAT Keep Alive Msg: $NOTIFY NAT Keep Alive Dest: $PROXY

Network Settings

SIP ToS/DiffServ Value: 0x68 SIP CoS Value: 3 [0-7]

RTP ToS/DiffServ Value: 0xb8 RTP CoS Value: 6 [0-7]

Network Jitter Level: high Jitter Buffer Adjustment: up and down

SIP Settings

SIP Transport: UDP SIP Port: 5061

SIP 100REL Enable: no EXT SIP Port:

Auth Resync-Reboot: yes SIP Proxy-Require:

SIP Remote-Party-ID: yes SIP GUID: no

SIP Debug Option: none RTP Log Intvl: 0

Restrict Source IP: no Referor Bye Delay: 4

Refer Target Bye Delay: 0 Referee Bye Delay: 0

Refer-To Target Contact: no Sticky 183: no

Auth INVITE: no Use Anonymous With RPID: yes

Use Local Addr In FROM: no

Proxy and Registration

Proxy: sipgate.at

Outbound Proxy: sipgate.at

Use Outbound Proxy: yes Use OB Proxy In Dialog: yes

Register: yes Make Call Without Reg: no

Register Expires: 3600 Ans Call Without Reg: no

Use DNS SRV: no DNS SRV Auto Prefix: no

Proxy Fallback Intvl: 3600 Proxy Redundancy Method: Normal

Subscriber Information

Display Name: User ID: 999999999999999

Password: ************* Use Auth ID: no

Auth ID: 99999999999999

Mini Certificate:

SRTP Private Key:

Audio Configuration

Preferred Codec: G711u Silence Supp Enable: no

Use Pref Codec Only: no Echo Canc Enable: yes

G729a Enable: yes Echo Canc Adapt Enable: yes

G723 Enable: yes Echo Supp Enable: yes

G726-16 Enable: yes FAX CED Detect Enable: yes

G726-24 Enable: yes FAX CNG Detect Enable: yes

G726-32 Enable: yes FAX Passthru Codec: G711u

G726-40 Enable: yes FAX Codec Symmetric: yes

DTMF Process INFO: yes FAX Passthru Method: NSE

DTMF Process AVT: yes DTMF Tx Method: Auto

DTMF Tx Mode: Strict DTMF Tx Strict Hold Off Time: 40

Release Unused Codec: yes FAX Process NSE: yes

Symmetric RTP: yes FAX Disable ECAN: no

Audio Dump Option1: none Audio Dump Option2: none

PSTN Line

Cisco SPA Configuration http://10.0.1.138/admin/voice/advanced

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Dial Plans

Dial Plan 1: (xx.)

Dial Plan 2: (xx.)

Dial Plan 3: (xx.)

Dial Plan 4: (xx.)

Dial Plan 5: (xx.)

Dial Plan 6: (xx.)

Dial Plan 7: (xx.)

Dial Plan 8: (S0<:0043999999999999@sipgate.at)

VoIP-To-PSTN Gateway Setup

VoIP-To-PSTN Gateway Enable: yes VoIP Caller Auth Method: none

VoIP PIN Max Retry: 3 One Stage Dialing: yes

Line 1 VoIP Caller DP: 1 VoIP Caller Default DP: 1

Line 1 Fallback DP: none

VoIP Caller ID Pattern:

VoIP Access List:

VoIP Caller 1 PIN: VoIP Caller 1 DP: 1

VoIP Caller 2 PIN: VoIP Caller 2 DP: 1

VoIP Caller 3 PIN: VoIP Caller 3 DP: 1

VoIP Caller 4 PIN: VoIP Caller 4 DP: 1

VoIP Caller 5 PIN: VoIP Caller 5 DP: 1

VoIP Caller 6 PIN: VoIP Caller 6 DP: 1

VoIP Caller 7 PIN: VoIP Caller 7 DP: 1

VoIP Caller 8 PIN: VoIP Caller 8 DP: 1

VoIP Users and Passwords (HTTP Authentication)

VoIP User 1 Auth ID: VoIP User 1 DP: 1

VoIP User 1 Password:

VoIP User 2 Auth ID: VoIP User 2 DP: 1

VoIP User 2 Password:

VoIP User 3 Auth ID: VoIP User 3 DP: 1

VoIP User 3 Password:

VoIP User 4 Auth ID: VoIP User 4 DP: 1

VoIP User 4 Password:

VoIP User 5 ID Auth ID: VoIP User 5 DP: 1

VoIP User 5 Password:

VoIP User 6 Auth ID: VoIP User 6 DP: 1

VoIP User 6 Password:

VoIP User 7 Auth ID: VoIP User 7 DP: 1

VoIP User 7 Password:

VoIP User 8 Auth ID: VoIP User 8 DP: 1

VoIP User 8 Password:

PSTN-To-VoIP Gateway Setup

PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none

PSTN Ring Thru Line 1: yes PSTN PIN Max Retry: 3

PSTN CID For VoIP CID: yes PSTN CID Number Prefix:

PSTN Caller Default DP: 8 Off Hook While Calling VoIP: no

Line 1 Signal Hook Flash To PSTN: Disabled PSTN CID Name Prefix:

PSTN Caller ID Pattern:

PSTN Access List:

PSTN Caller 1 PIN: PSTN Caller 1 DP: 1

PSTN Caller 2 PIN: PSTN Caller 2 DP: 1

PSTN Caller 3 PIN: PSTN Caller 3 DP: 1

PSTN Caller 4 PIN: PSTN Caller 4 DP: 1

PSTN Caller 5 PIN: PSTN Caller 5 DP: 1

PSTN Caller 6 PIN: PSTN Caller 6 DP: 1

PSTN Caller 7 PIN: PSTN Caller 7 DP: 1

PSTN Caller 8 PIN: PSTN Caller 8 DP: 1

FXO Timer Values (sec)

VoIP Answer Delay: 0 VoIP PIN Digit Timeout: 10

PSTN Answer Delay: 0 PSTN PIN Digit Timeout: 10

PSTN-To-VoIP Call Max Dur: 0 PSTN Ring Thru Delay: 1

VoIP-To-PSTN Call Max Dur: 0 PSTN Ring Thru CWT Delay: 3

VoIP DLG Refresh Intvl: 0 PSTN Ring Timeout: 5

Cisco SPA Configuration http://10.0.1.138/admin/voice/advanced

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User Login basic | advanced

Copyright © 1992-2009 Cisco Systems, Inc. All Rights Reserved.

PSTN Dialing Delay: 1 PSTN Dial Digit Len: .1/.1

PSTN Hook Flash Len: .25

PSTN Disconnect Detection

Detect CPC: yes Detect Polarity Reversal: yes

Detect PSTN Long Silence: no Detect VoIP Long Silence: no

PSTN Long Silence Duration: 30 VoIP Long Silence Duration: 30

PSTN Silence Threshold: medium Min CPC Duration: 0.2

Detect Disconnect Tone: yes

Disconnect Tone: 480@-30,620@-30;4(.25/.25/1+2)

International Control

FXO Port Impedance: 600 Ring Frequency Min: 10

Dtmf Playback Level: -7.3 Dtmf Playback Twist: 1.3

SPA To PSTN Gain: 0 Ring Frequency Max: 90

PSTN To SPA Gain: 0 Ring Validation Time: 256 ms

Tip/Ring Voltage Adjust: 3.5 V Ring Indication Delay: 512 ms

Operational Loop Current Min: 10 mA Ring Timeout: 640 ms

On-Hook Speed: Less than 0.5 ms Ring Threshold: 13.5-16.5 Vrms

Current Limiting Enable: no Ringer Impedance: High (Normal)

Line-In-Use Voltage: 30

Cisco SPA Configuration http://10.0.1.138/admin/voice/advanced

3 von 3 03.03.2013 10:44

99hwittenb

[a123a123a123]

Hello 99hwittenb!

update: Using my DECT Phone I really had to dial to 0043999999999@sipgate.at to get a dial tone! This previously wasn't clear to me!

Then I entered a phone number and SPA3102 called out using PSTN!

However:

Using X-Lite calling 0043999999999@sipgate.at to get a dial tone didn't work, Call did not even start.

Any idea to use this functionality with X-Lite?

KR H.

[a123a123a123]

Hello 99hwittenb!

Hello 99hwittenb!

I tried to configure a dial-out pin according to the following guideline:

...

2. SPA3102 Web-UI > Voice tab > PSTN Line tab > PSTN-To-VoIP Gateway Setup

a. PSTN Caller Auth Method: PIN

b. PSTN Caller 1 PIN: 123

Use something more secure than my example, else someone could run up your account.

    FXO Timer Values (sec) >

c. PSTN Answer Delay: 1

   [This causes the SPA3102 to pick up the call in 1 second instead of making you wait the default 16 seconds]

3. Click Submit All Changes

4.  After the SPA3102 has booted, verify that both Line 1 Status and PSTN  LIne Status have a Registered Registration State when viewed in the  SPA3102 Web-UI > Voice tab > Info tab

However I figured out the following:

calling regular PSTN Nr 0043111111 ==> PSTN 2 VOIP Gateway ring at VOIP Phone (OK)

calling 0043999999@sipgate.at ==> VOIP 2 PSTN Gateway gives  dial tone (tone OK, but no pin doesn't get accepted - dialout after tone  is still possible!)

calling VOIP-PSTN Nr 0043999999 ==> VOIP 2 PSTN Gateway gives  dial tone (NOK, VOIP Call should be stablished, moreover no pin  available - dialout after tone is possible!)

Question 1:

Do you have an idea, why pin is not working?

Question 2: using same VOIP account for dial out and receive calls is not as desired, is the following setting possible?

- calling regular PSTN Nr 0043111111 ==> PSTN 2 VOIP Gateway using account 0043999999@sipgate.at

- calling VOIP-PSTN Nr 0043999999 should ring for normal calls on VOIP Phone (mapping is done in fritz-box)

- calling with different account 004388888888@sipgate.at ==> VOIP 2 PSTN Gateway gives dial  tone + prompts for pin

- calling VOIP-PSTN-Nr 004388888888 ==> VOIP 2 PSTN Gateway gives dial tone + prompts for pin

Thank you for your reply!

KR H.

[Howard Wittenberg]

I am not sure what you mean by the SPA3102's internal bridge -- "( I'm behind the Fritzbox ADSL Modem, however I use the SPA3102's internal bridge functionality for the ethernet port)"

The SPA3102 should be attached to the network via the "Internet" port (jack).  I believe the SPA3102 design assumption is that calls will come in and go out thru the "Internet" port and not thru the "Ethernet" port.  The "Ethernet" port is for the router function of the SPA3102 to attach computers or other devices that you wish to place downstream from the SPA3102.  With my testing I never could get calls to work over the "ethernet" port.  If you are passing the call thru the "Ethernet" port that could be causing a problem.

I don't understand the capabilities of the "Fritzbox".  It is not sold or widely used in the U.S. and I am not sure of the difference between the Fritzbox and Xlite or exactly how you are using either one.  If you have the SPA3102 PSTN Line Tab registered to an account at sipgate.at and you call that account from Fritzbox or from XLite I would think they would work the same.

I would get things working properly without a pin.  Then you can start introducing pin authentication.  As you know, the voip-to-pstn gateway setting is for incoming voip calls to the SPA3102 to be bridged to the pstn line, the pstn-to-voip gateway is for incoming pstn line calls to the SPA3102 to be bridged to the voip line.  Each has separate pin authentication settings.

For your question about a different outgoing voip account from an incoming voip account, you can setup a different "Gateway" voip account to be used for outgoing pstn-to-voip calls.  The account is for outgoing calls only and the voip provider must accept outgoing calls without registration.  You setup the "Gateway" account on the Line 1 Tab.  On the PSTN Line Tab dial plan (PSTN Caller Default DP:) you reference the gateway account.  If you have the account setup in Gateway 1 the dial plan would be something like (xx.<:@gw1>).  The format for setting up a gateway account is

Gateway 1: userid@sip_proxy

GW1 AuthID: userid

GW1 Password: password

GW1 NAT Mapping Enable: Yes or No

NAT Mapping Enable affects whether the sip signalling uses the local (internal) network ip address or uses the external network ip address.

[a123a123a123]

Hello 99hwittenb!

Hello 99hwittenb!

Hello 99hwittenb!

Thank you for your help!

> The SPA3102 should be attached to the network via the "Internet" port (jack).

You are right, I was typing the wrong port in my mail. It was connected through the internet port to the fritz.box.

> If you have the SPA3102 PSTN Line Tab registered to an account at sipgate.at

> and you call that account from Fritzbox or from XLite I would think they would work the same.

Right, but through fritz.box it finally worked. It was also working when I called the voip-pstn-nr 0043999999 associated with the voip account.

X Lite I got an error.

> I would get things working properly without a pin. 

> Then you can start introducing pin authentication. 

> As you know, the voip-to-pstn gateway setting is for incoming

> voip calls to the SPA3102 to be bridged to the pstn line

That's what I need.

VOIP to PSTN works, but I don't want to use it without pin.

To switch it on I made the following settings: Auth Method: PIN and Caller 1 PIN: 9999

I tried introducing both pins (voip-to-pstn and pstn-to-voip) in separate trial runs, none of them worked.

Are there any other settings to consider when I introduce a pin?

> ... different "Gateway" ...

Sounds rather interesting, I'll try to figure it out when I was able to introduce pin successfully!

The  SPA3102 should be attached to the network via the "Internet" port  (jack). - See more at:  https://supportforums.cisco.com/message/3871690#sthash.vpVmuHAZ.dpuf

Thanks for your reply!

KR H.

[Howard Wittenberg]

I reviewed the pin documentation and tried testing to see in what circumstance the pin didn't work.  It always challenges you with 3 short beeps.  After the 3 beeps you enter the pin and follow it with #.  If it accepts the pin the SPA3102 follows with a dial tone, if not you get a fast busy tone.  It tried a number of times, when I tried it always presented the 3 beeps.  Sometimes it rejected the pin I entered.

The other setting to consider when introducing pins and for that matter responding to a dial tone is the setting for sending DTMF tones over the network. You are receiving DTMF tones on the SPA3102 so it may be a setting on your Frtiz box.  The preferred setting on the SPA3102 for sending dtmf tones is usually rfc2833 which Cisco/Linksys calls AVT.  On the SPA3102 the setting is DTMF Transmit Method: InBand/AUTO/AVT/INFO.  On the SPA3102 there is also DTMF Tx Mode: Normal/Strict.  This has to do with the length of the tone.  I favor normal.

This is what the Admin manual says about PIN authentication:

You also can enable PIN authentication. In this case, the VoIP caller is prompted to

enter a PIN number after the SPA3102 answers the call. The PIN number must end

with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to

eight VoIP caller PIN numbers can be configured on the SPA3102. A dial plan can

be selected for each PIN number. If the caller enters a wrong PIN or the SPA3102

times out waiting for more PIN digits, the SPA3102 tears down the call

immediately with a BYE request

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf

In addition to Pin authentication for voip-to-pstn bridging there is also http digest authentication.  This is a superior method of authentication but can not be done with a regular call from a voip provider.  In this method of communication the authentication is done in the sip invite by passing an encrypted password and is similiar to the authentication done by voip providers.  To use http digest authentication you setup the SPA3102 as an outgoing trunk in the calling phone or adapter.  You setup an Auth ID (UserID) and Password which is sent after a challenge in the caller's sip INVITE.  When using http digest authentication you typically setup direct ip dialing and do not go thru a voip provider.  Usually http digest authentication uses one stage dialing and does not pass dtmf digits over the network which is a reliability advantage.

[a123a123a123]

Hello 99hwittenb!

Thank you very much for the in depth explainations regarding authentication!

As I mentioned above unfortunately I never got 3 short beeps to ask for a pin.

However I will review all settings next weekend when I'm home again.

I'll let you know if I was able to manage it!

Thanks a lot!

KR H.

[a123a123a123]

hello 99hwittenb!

> As I mentioned above unfortunately I never got 3 short beeps to ask for a pin.

When I changed for 2 stage dialling then it worked.

Gateway:

(xx.<:@gw1>).    set as dial plan 7 - PSTN default DP: 7

Gateway 1: 12345@MySipProvider.at (= is a valid voip number)

GW1 AuthID: 12345

GW1 Password: password

GW1 NAT Mapping Enable: Yes or No  (no matter what I set, same results)

When I call:

calling phone get following signal:

tüt tuuuuuuuuuuu tüt tüt tüt tüt

==> VOIP account does not ring, neither in DECT nor in X-Lite

Isn't there a necessity to invoke my former default dial plan Nr 8?

(S0<:00431234556789@mySipProvider.at)

KR H.

[Howard Wittenberg]

> As I mentioned above unfortunately I never got 3 short beeps to ask for a pin.

When I changed for 2 stage dialling then it worked.

Good.  That was for the voip-to-pstn gateway calling.

For the pstn-to-voip outbound calling if you wish to use a different voip account than setup on the PSTN Line Tab you can setup the account on one of the SPA3102 gateway configurations.  The gateway configurations are setup on the Line 1 Tab.

For the dial plan to access the different account (xx.<:@gw1>) is the way you do it if you wish a dial tone.  If you wish to automatically call a specific number you would have a dial plan something like this ... (S0<:00431234567890@gw1>).

Note that the PSTN Caller Default DP: is used when you do not have PIN authentication.  When you setup PIN authentication the Dial Plan number used is the PSTN Caller 1 DP: on the same line as the PIN number. It is different.  Get it working without a pin and then introduce the pin authentication.

If the above dial plans do not work you have some other error occuring.  Could be DNS but more than likely it is something to do with DTMF tones over the network.  A sip debug trace will tell what the problem is.  Solving it may be more challenging.

The sound you described is a reorder error tone.

A dial plan (S0:0043123456789@mySipProvider.at) is automatically dialing a sip uri.  You could use that to call a specific number at an account if the sip provider accepts incoming sip uri calls and does not challenge for a userid and password.  If calling the number involves authentication you need to use the configuration setup on the PSTN Line Tab or you need to specifiy a configuration setup on one of the gateway accounts.

[a123a123a123]

Hello 998hwittenb!

Thank you for your instructions to the gateway!

Finally I could successfully install a Y-Calbe to my Fritz-box to enable phone and internet directly with the Fritz.box.

As a consequence I can't connect any other phone devices to my two other FXS connectors, because the Fritz.box catches it.

Therefore I won't be able to use SPA 3102 any loger.

In any case I'd say thank you many times for answering my beginner questions!

KR H.