Howto:Swisscom Business Connect - SIP Trunk - SIP Provider Compatibility Test: Difference between revisions

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(New page: '''Innovaphone Compatibility Test Report''' == Summary == '''SIP Provider: Swisscom''' <!-- The provider supports all required innovaphone features and is therefore qualified as [[Howto...)
 
 
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== Summary ==
== Summary ==


'''SIP Provider: Swisscom'''
'''SIP Provider: swisscom Business Connect'''


<!-- The provider supports all required innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]]. -->
The provider supports all mandatory innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].


That being said, the provider has achieved 86,34% of all possible test points (139 of 161). For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description_v11#|Test Description V11 and later]]


...


That being said, the provider has achieved x% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#|Test Description]]
'''Please Note, that some features are restricted or not supported by swisscom'''
<!-- Mention all important tests that were not passed in the summary.  
* Fax over IP is on principle not supported by swisscom
** The T.38 feature is not available
** Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!
* Only G.711a is supported, no other codecs are supported
* Overlapped Sending is not supported
* NAT Detection is not supported
* Redundancy Mechanism are not supported
* SIP over TCP is not supported


E.g. in case that the provider doesn't support Reverse Media-Negotiation, mention in the summary that media relay and an exclusive coder setting must be configured:
* Incoming Early Media Channel is not supported by default and must be requestet to swisscom.
* CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work.


Since the provider doesn't support Reverse Media negotiation, media relay and an exclusive coder setting must be configured. Opposed to a SIP trunk not needing Media-Relay, the transport of all RTP packets by the gateway will result in a higher CPU load for a call. As a result, the amount of concurrent calls is considerably lower compared to a SIP-Provider that doesn't require Media-Relay.
-->


<!-- in case of customer testimonial, please make sure that the fact that it is not tested by innovaphone is also mentioned in the summary-->
'''swisscom doesn't support "Reverse Media negotiation". Because of this, "Media-Relay" and an "Exclusive Coder Setting" must be configured.''' In contrast to a SIP trunk which does not need Media-Relay, the transport of all RTP packets across the gateway will result in a higher CPU load.


* Features:
In order to use some Features like[[Howto:How_does_CLIP_no_screening_work | Clip no screening]], some [[Howto:Swisscom_-_SIP_Trunk_-_SIP_Provider_Compatibility_Test#Configuration|configuration]] must be done.


** Direct Dial In
Supported Features:
** Fax over IP (T.38)
* Direct Dial In
** DTMF
* DTMF
* CGPN can be suppressed
* CLIP No Screening
* Reverse Media Negotiation
* Hold/Retrieve
* Blind Transfer


* Supported Codecs by the provider
Supported Codecs:
** G711
* G.711a only
** G729
** G723
** G726
** T.38 UDP


== Current test state ==
== Current test state ==


{{Template:Compat Status "planned"}}
<!-- {{Template:Compat Status "in progress"}}-->
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat Status "planned"}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"referral_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
<!-- {{Template:Compat_Status_"engineered_prod."|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}} -->
Line 44: Line 51:
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->
<!-- {{Template:Compat_Status_"referral_prod."-no-certificate}} -->


<!-- Testing of this product has been finalized January 1st, 1970. -->
{{Template:Compat Status "certified"|certificate=Swisscom_Business_Connect_-_SIP_Provider_-_product-cert.pdf}}


<internal>
Testing of this product has been finalized February 2015.
Beim Abschluss des Tests (egal ob gut, schlecht oder abgebrochen) <strong>bitte Nachricht an ckl</strong>!
</internal>


== Testing Enviroment ==
== Testing Enviroment ==


[[Image:SIPProviderTestTopology1.PNG]]
[[Image:swisscomSIPProviderTestTopology1.png]]


This scenario describes a setup where the PBX and phones are in a private network.  
This scenario describes a setup where the PBX and phones are in a private network.


There are 3 major configuration variants of the SIP trunk, which one is used depends on the test results. The variants are:
The swisscom is providing an own Gateway ("Connect Box" or "Patton Smart Node") which must be located in the customers private network. This Gateway has two connections:
* One connection to swisscom
* One connection to the PBX
Because of this, the SIP Trunk does not connect directly to the internet, but to the swisscom Gateway ("Connect Box" or "Patton Smart Node").


* the SIP trunk is configured without Media Relay and without exclusive coder. This is the case if all tests were successful
The SIP trunk must be configured with "Media Relay", "exclusive G.711a" and "NO T.38".
* the SIP trunk is configured with Media Relay and STUN but without exclusive coder. This is the case when the test for "NAT Traversal" fails
* the SIP trunk is configured with Media Relay and exclusive coder. This is the case when the test for "Reverse Media Negotiation" fails


The test scenario should describe which SIP trunk configuration is needed.
[[Howto:Swisscom_-_SIP_Trunk_-_SIP_Provider_Compatibility_Test#Configuration|See configuration]]


== Test Results ==
== Test Results ==


For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description|SIP Interop Test Description]]. Bold lines in the test results indicate a KO-criteria.
For more information on the test procedure, please read the following wiki article: [[Howto:SIP_Interop_Test_Description_v11#|Test Description V11 and later]]. Bold lines in the test results indicate a KO-criteria.


{{FIXME|reason=Note: Possible test results are Ok, Nok, Not tested, N/A(not available). Ok, Nok, N/A are self explanatory. "Not tested" is more a temporary note - if its still in the test after finishing, then the summary should explain why the feature was not tested. Remove this note in the final report}}
(*1) To use Incoming Early Media Channel, this feature must be requested to swisscom. Per default, it's not possible.
 
(*2) Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!
 
(*3) CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work in this case.  


=== Basic Call ===
=== Basic Call ===
Line 77: Line 87:
|----
|----
|'''call using g711a'''
|'''call using g711a'''
|
|'''OK'''
|----
|----
|'''call using g711u'''
|'''call using g711u'''
|
|'''NOK'''
|----
|----
|call using g723
|call using g723
|
|NA
|----
|----
|call using g729
|call using g729
|
|NA
|----
|----
|call using g722
|call using g722
|
|NA
|----
|----
|Overlapped sending
|Overlapped sending
|
|NA
|----
|----
|'''early media channel'''
|'''early media channel outgoing'''
|
|'''OK'''
|----
|early media channel incoming
|OK/NOK (*1)
|----
|----
|Fax using T.38
|Fax using T.38
|
|NA
|----
|----
|T.38 Transcoding by the provider
|T.38 Transcoding by the provider
|
|NA
|----
|Fax using G.711
|OK (*2)
|----
|----
|Reverse Media Negotiation
|Reverse Media Negotiation
|
|OK
|----
|----
|CGPN can be suppressed
|CGPN can be suppressed
|
|OK
|----
|----
|CLIP no screening
|CLIP no screening
|
|OK (*3)
|----
|----
|'''Long time call possible(>30 min)'''
|'''Long time call possible(>30 min)'''
|
|'''OK'''
|----
|----
|'''External Transfer'''
|'''External Transfer'''
|
|'''OK'''
|----
|----
|NAT Detection
|NAT Detection
|
|NA
|----
|----
|Redundancy
|Redundancy
|
|NA
|----
|----
|SIP over TCP
|SIP over TCP
|
|NA
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|
|'''OK'''
|}
|}


Line 138: Line 154:
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|
|'''OK'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|
|'''OK'''
|----
|----
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|'''Loop In call(Innovaphone -> Provider -> Innovaphone)'''
|
|'''OK (*3)'''
|}
|}


Line 154: Line 170:
|----
|----
|'''DTMF tones sent correctly via RTP-events(RFC 2833)'''
|'''DTMF tones sent correctly via RTP-events(RFC 2833)'''
|
|'''OK'''
|----
|----
|DTMF tones sent correctly via SIP-Info
|DTMF tones sent correctly via SIP-Info
|
|OK
|----
|----
|'''DTMF tones received correctly via RTP-events(RFC 2833)'''
|'''DTMF tones received correctly via RTP-events(RFC 2833)'''
|
|'''OK'''
|}
|}


Line 170: Line 186:
|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|
|'''OK'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|
|OK
|}
|}


Line 183: Line 199:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|OK
|}
|}


Line 197: Line 213:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|
|OK
|}
|}


Line 228: Line 244:
|----
|----
|'''Call can be transferred'''
|'''Call can be transferred'''
|
|'''OK'''
|----
|----
|Held end hears music on hold or dialling tone
|Held end hears music on hold or dialling tone
|
|OK
|----
|----
|'''Call returns to transferring device if the third'''  
|'''Call returns to transferring device if the third'''  
'''Endpoint is not available'''
'''Endpoint is not available'''
|
|'''OK'''
|}
|}


Line 246: Line 262:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|
|OK
|}
|}


Line 277: Line 293:
|----
|----
|Call can be transferred
|Call can be transferred
|
|OK
|----
|----
|Held end hears dialling tone
|Held end hears dialling tone
|
|OK
|}
|}


Line 290: Line 306:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|----
|----
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|inno1 calls PSTN-phone. inno1 transfers to inno2.
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|PSTN-phone calls inno1. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|}
|}


Line 315: Line 331:
|----
|----
|'''Call can be forward'''
|'''Call can be forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 328: Line 344:
|----
|----
|'''Call can be transferred or forward'''
|'''Call can be transferred or forward'''
|
|'''OK'''
|----
|----
|'''Held end hears dialling tone'''
|'''Held end hears dialling tone'''
|
|'''OK'''
|}
|}


Line 341: Line 357:
|----
|----
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|inno1 calls inno2. inno2 transfers to PSTN-phone.
|
|OK
|----
|----
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|inno1 calls PSTN-phone. PSTN-phone transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to inno2.
|PSTN-phone calls inno1. inno1 transfers to inno2.
|
|OK
|----
|----
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2.
|
|OK
|}
|}


Line 360: Line 376:
|----
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Caller can make a call to a Broadcast Group'''
|
|'''OK'''
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Caller can make a call to a Waiting Queue'''
|
|'''OK'''
|----
|----
|'''Announcement if nobody picks up the call'''
|'''Announcement if nobody picks up the call'''
|
|'''OK'''
|}
|}


Line 373: Line 389:
===Firmware version===
===Firmware version===


All innovaphone devices use Vx build xx-xxxxx as firmware.
All innovaphone devices use V11 build 11.0828 as firmware during test.


=== SIP - Trunk ===
=== SIP - Trunk ===
Following configuration needs to be done in the SIP Gateway Interface settings:


* 1: Enter your phonenumber and private IP address of swisscom Gateway ("Connect Box" or "Patton Smart Node") with port 5062!
* 2: Enter credendtials
* 3: Select G.711a exclusive
* 4: Disable T.38 and Enable Media Relay
* 5: Disable encryption
* 6: Use "CGPN in user part of URI" in "From Header" and "Fixed AOR" in "Identity Header"
[[Image:swisscomSIPProviderTestConfig1.png]]


=== Number Mapping ===
=== Number Mapping ===
Following configuration needs to be done in the SIP Gateway Number Mappings:


* 1: Add "00" to incoming CGPNs without a leading "0" because international numbers are coming in like "49703173009".
* 2: Because swisscom is using the full national number for incoming calls, you need to cut off the digits before an extension. In example, 04333693xx will be translated to xx.
* 3: In case of call forwardings, you need to cut off the single leading "0" (national calls) and translate a leading "000" (international calls) to "+" in the CGPN Out map.
* 4: In the CGPN Out map, you also need to translate the extensions to the national number format. In example, xx will be translated to 04333693xx.
[[Image:swisscomSIPProviderTestConfig2.png]]


=== Route Settings ===
=== Route Settings ===
A default Route Setup is used:
* 1: Force Enblock Setting is required for outgoing calls.


[[Image:swisscomSIPProviderTestConfig3.png]]


=== Media Relay ===
=== Fax ===
'''swisscom does not support any Fax communications.''' Because of this, T.38 can not be used.


If you want to use Fax even so, you need to set T.38 to G.711 translation (Audio Fax support, available from V11). This works in a lot of cases where you have no problems with QoS. '''But it's not guaranteed and swisscom gives no support in case of problems!''' 


=== Fax ===
Example to set T.38 to G.711 translation (Audio Fax support) on the Faxserver interface:


[[Image:swisscomSIPProviderTestConfig4.png]]


[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]

Latest revision as of 13:26, 2 March 2015

Innovaphone Compatibility Test Report

Summary

SIP Provider: swisscom Business Connect

The provider supports all mandatory innovaphone features and is therefore qualified as recommended SIP Provider.

That being said, the provider has achieved 86,34% of all possible test points (139 of 161). For more information on the test rating, please refer to Test Description V11 and later


Please Note, that some features are restricted or not supported by swisscom

  • Fax over IP is on principle not supported by swisscom
    • The T.38 feature is not available
    • Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!
  • Only G.711a is supported, no other codecs are supported
  • Overlapped Sending is not supported
  • NAT Detection is not supported
  • Redundancy Mechanism are not supported
  • SIP over TCP is not supported
  • Incoming Early Media Channel is not supported by default and must be requestet to swisscom.
  • CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work.


swisscom doesn't support "Reverse Media negotiation". Because of this, "Media-Relay" and an "Exclusive Coder Setting" must be configured. In contrast to a SIP trunk which does not need Media-Relay, the transport of all RTP packets across the gateway will result in a higher CPU load.

In order to use some Features like Clip no screening, some configuration must be done.

Supported Features:

  • Direct Dial In
  • DTMF
  • CGPN can be suppressed
  • CLIP No Screening
  • Reverse Media Negotiation
  • Hold/Retrieve
  • Blind Transfer

Supported Codecs:

  • G.711a only

Current test state

Recprod.PNG The tests for this product have been completed and it has been approved as a recommended product (Certification document).

Testing of this product has been finalized February 2015.

Testing Enviroment

SwisscomSIPProviderTestTopology1.png

This scenario describes a setup where the PBX and phones are in a private network.

The swisscom is providing an own Gateway ("Connect Box" or "Patton Smart Node") which must be located in the customers private network. This Gateway has two connections:

  • One connection to swisscom
  • One connection to the PBX

Because of this, the SIP Trunk does not connect directly to the internet, but to the swisscom Gateway ("Connect Box" or "Patton Smart Node").

The SIP trunk must be configured with "Media Relay", "exclusive G.711a" and "NO T.38".

See configuration

Test Results

For more information on the test procedure, please read the following wiki article: Test Description V11 and later. Bold lines in the test results indicate a KO-criteria.

(*1) To use Incoming Early Media Channel, this feature must be requested to swisscom. Per default, it's not possible.

(*2) Fax via G.711 (T.38 to G.711 / Audio Fax support) is not supported by swisscom but should work in most scenarios. This Feature is not guaranteed!

(*3) CLIP no screening does not work for "Trunk loop Connections". This means, if you dial out to the PSTN and back to your own Trunk Number, CLNS does not work in this case.

Basic Call

Tested feature Result
call using g711a OK
call using g711u NOK
call using g723 NA
call using g729 NA
call using g722 NA
Overlapped sending NA
early media channel outgoing OK
early media channel incoming OK/NOK (*1)
Fax using T.38 NA
T.38 Transcoding by the provider NA
Fax using G.711 OK (*2)
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening OK (*3)
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection NA
Redundancy NA
SIP over TCP NA
Voice Quality OK? OK

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone) OK
Outbound(Innovaphone -> Provider) OK
Loop In call(Innovaphone -> Provider -> Innovaphone) OK (*3)

DTMF

Tested feature Result
DTMF tones sent correctly via RTP-events(RFC 2833) OK
DTMF tones sent correctly via SIP-Info OK
DTMF tones received correctly via RTP-events(RFC 2833) OK

Hold/Retrieve

Tested feature Result
Call can be put on hold OK
Held end hears music on hold / announcement from PBX OK

Transfer with consultation

Tested feature Result
Call can be transferred OK
Held end hears music on hold OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK OK

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred OK
Held end hears music on hold or dialling tone OK
Call returns to transferring device if the third

Endpoint is not available

OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK OK

Blind Transfer

Tested feature Result
Call can be transferred OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK

CFU / CFB Transfer

Tested feature Result
Call can be forward OK
Held end hears dialling tone OK

CFNR / Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group OK
Caller can make a call to a Waiting Queue OK
Announcement if nobody picks up the call OK

Configuration

Firmware version

All innovaphone devices use V11 build 11.0828 as firmware during test.

SIP - Trunk

Following configuration needs to be done in the SIP Gateway Interface settings:

  • 1: Enter your phonenumber and private IP address of swisscom Gateway ("Connect Box" or "Patton Smart Node") with port 5062!
  • 2: Enter credendtials
  • 3: Select G.711a exclusive
  • 4: Disable T.38 and Enable Media Relay
  • 5: Disable encryption
  • 6: Use "CGPN in user part of URI" in "From Header" and "Fixed AOR" in "Identity Header"

SwisscomSIPProviderTestConfig1.png

Number Mapping

Following configuration needs to be done in the SIP Gateway Number Mappings:

  • 1: Add "00" to incoming CGPNs without a leading "0" because international numbers are coming in like "49703173009".
  • 2: Because swisscom is using the full national number for incoming calls, you need to cut off the digits before an extension. In example, 04333693xx will be translated to xx.
  • 3: In case of call forwardings, you need to cut off the single leading "0" (national calls) and translate a leading "000" (international calls) to "+" in the CGPN Out map.
  • 4: In the CGPN Out map, you also need to translate the extensions to the national number format. In example, xx will be translated to 04333693xx.

SwisscomSIPProviderTestConfig2.png

Route Settings

A default Route Setup is used:

  • 1: Force Enblock Setting is required for outgoing calls.

SwisscomSIPProviderTestConfig3.png

Fax

swisscom does not support any Fax communications. Because of this, T.38 can not be used.

If you want to use Fax even so, you need to set T.38 to G.711 translation (Audio Fax support, available from V11). This works in a lot of cases where you have no problems with QoS. But it's not guaranteed and swisscom gives no support in case of problems!

Example to set T.38 to G.711 translation (Audio Fax support) on the Faxserver interface:

SwisscomSIPProviderTestConfig4.png

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