Elastix PBX (Asterisk) + My PME Office

Hello, I am looking for information to add a Swisscom “SIP Trunk” under Elastix (PBX based on Asterisk), but for the moment I have not yet succeeded.

I have a My SME Office L+ router subscription Centro Business PSB4212N

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Here are my current configs that I found here:

__Output Parameters*

Trunk Name: Swisscom

PEER details:

user=+4121XXXXXXX
type=peer
srvlookup=yes
secret=password
outboundproxy=bc1.ims.swisscom.ch
nat=yes
insecure=prompt,port
host=swisscom.ch
fromuser=+4121XXXXXXX
fromdomain=swisscom.ch
dtmfmode=auto
defaultuser=+4121XXXXXXX
canreinvite=no
disallow=all
allow=alaw&ulaw&g729&gsm&slinear&ulaw

Input Parameters

User Context: +4121XXXXXXX

User Details:

type=peer
host=bc1.ims.swisscom.ch
fromdomain=swisscom.ch
disallow=all
allow=alaw&ulaw&g729&gsm&slinear&ulaw

Registration

Recording channel:

+4121XXXXXXX@swisscom.ch:+4121XXXXXXX:+4121XXXXXXX@swisscom.ch/+4121XXXXXXX

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If it helps in the analysis/research, I carried out packet captures with Wireshark + Swisscom “Business telephony” app at the time of a call to a 079 mobile and I obtained the following results:

Request: INVITE sip:079XXXXXXX@swisscom.ch

Request: ACK sip:079XXXXXXX@swisscom.ch

Request: NOTIFY sip:+4121XXXXXXX-01@192.168.XX.XX:5075;transport=tcp

Request: PRACK sip:sgc_c@195.186.128.4:5075;transport=tcp

Request: BYE sip:+4121XXXXXXX-01@192.168.XX.XX:5075;transport=tcp

Wikipedia Info

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Here is the output in the Elastix CLI (ssh) at the time of registration:

[2015-12-04 11:14:25] NOTICE[2478]: chan_sip.c:15067 sip_reg_timeout: – Registration for ‘+4121XXXXXXX@swisscom. ch’ timed out, trying again (Attempt #5)
Really destroying SIP dialog ‘305e3940689583541cfb3a9843631eb4@127.0.0.1’ Method: REGISTER
Retransmitting #1 (no NAT) to 193.222.73.227:5060:
REGISTER sip:swisscom.ch SIP/2.0
Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK3b33d7ce
Max-Forwards: 70
From: <sip:4121XXXXXXX@Anonym.ch>;tag=as66323714
To: <sip:4121XXXXXXX@Anonym.ch>
Call-ID: 305e3940689583541cfb3a9843631eb4@127.0.0.1
CSeq: 106 REGISTER
User-Agent: FPBX-2.11.0(11.13.0)
Expires: 120
Contact: <sip:4121XXXXXXX@192.168.10.52:5060>
Content-Length: 0

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Has anyone ever had this kind of configuration on a PBX like Asterisk, Elastix, Trixbox, etc.? or any idea to fix this problem? If yes, can you please share?

P.S: I have already called the Swisscom hotline but they don’t seem to be too interested in giving this kind of information.

Thanks in advance,

Best regards

:smileyvery-happy:

Show original language (French)

Dear Customer,

We thank you for your request. Indeed, we do not provide this information. We only have our partners who do such installations. If you want to have a solution with IP telephony with PBX, we recommend the Business Connect product, more detailed explanations of which can be found here:

[https://www.swisscom.ch/fr/business/pme/reseau-fixe/telephonie-avancee/business-connect.html](https://www.swisscom.ch/fr/business/pme/reseau- landline/advanced-telephony/business-connect.html)

In this case, it will no longer be a MY PME Office subscription but a Business Internet subscription.

We also remain at your disposal if necessary on 0800 055 055.

Best regards

Stefanie.C

Show original language (French)
22 days later

Hello Switzerland,

By feeling, I would see the following configuration:

__Output Parameters*

Trunk Name: Swisscom

PEER details:

user=+4121XXXXXXX
type=peer
srvlookup=yes
secret=+4121XXXXXXX (can be changed in the Centro Business GUI, under “Settings>VoIP>Change>Authentication Password”)
outboundproxy=192.168.1.1 (or the Centro Business LAN IP address if changed)
nat=yes
insecure=prompt,port
host=192.168.1.1 (or the Centro Business LAN IP address if changed)
fromuser=+4121XXXXXXX
fromdomain=192.168.1.1 (or the Centro Business LAN IP address if changed)
dtmfmode=Inband
defaultuser=+4121XXXXXXX
canreinvite=no
disallow=all
allow=alaw&ulaw&g729&gsm&slinear&ulaw

Input Parameters

User Context: +4121XXXXXXX

User Details:

type=peer
host=192.168.1.1 (or the Centro Business LAN IP address if changed)
fromdomain=192.168.1.1 (or the Centro Business LAN IP address if changed)
disallow=all
allow=alaw&ulaw&g729&gsm&slinear&ulaw

Show original language (French)
8 days later

Hello Krimskell, thank you for your response.

I just tested your configuration but I still have the same problem, so it is not possible to register the SIP Trunk.

Do you have an idea?

Thank you in advance for your response.

Best regards

:smileyhappy:

Show original language (French)
3 months later

Hello Switzerland,

I also have an Asterisk exchange, after 1 hour of support with Swisscom. I was unable to speak with an expert to transfer the technical data to me.

Does My PME Office use the standard SIP protocol? Did you manage to connect your Asterisk?

Best regards,

Laurent

Show original language (French)
15 days later

@lmu77 Hello, sorry I only saw the alerts for your messages now.. Have you managed to move forward on your side?

Would you be kind enough to share your information?

What configs do you use with Swisscom and Asterisk to connect with “My PME Office”?

Thank you in advance for your response,

Best regards.

Show original language (French)
5 days later
5 months later

Good morning,

I tried the configuration, but without success.

Can you share the detailed and complete configuration.

Thank you in advance and have a good day.

Freddy

Show original language (French)
7 months later
5 days later

Bonjour, voici la solution:

General Settings:

Trunk Name: SwisscomVoIPviaMyPMEOffice
Outbound Caller ID: “XYZ”<+4121XXXXXXX>
CID Options: Allow Any CID
Maximum Channels: 2

Dialed Number Manipulation Rules:

( prepend ) + prefix | match pattern

Outgoing Settings:

Trunk Name: swisscom

PEER Details:

type=friend
context=from-pstn
host=192.168.1.1
disallow=all
allow=alaw&ulaw
qualify=yes
username=+4121XXXXXXX

secret=MotdePasse123
nat=yes

Register String:

+4121XXXXXXX:MotdePasse123@192.168.1.1

P.S: Très important, le “Caller ID” doit obligatoirement être renseigné (juste)!

Meilleures Salutations

😃

2 months later

Good morning,

Asterisk can work with My-PME-Office and other products using the CentroBusiness2.0 router and even several home routers. However, we are systematically talking here about solutions which are not designed to support a PBX and therefore Swisscom does not support possible requests and problems related to this.

There is a solution allowing you to have Asterisk connected via SIP to Swisscom “legally”. It’s called ESIP, it’s a Large Business product that can also be obtained from SMEs. It involves some effort and knowledge but it is entirely possible even if some say not because they think mass market and Smart-Buesiness-Connect.

The price of the product also makes it very relevant compared to other services and suppliers.

Yours sincerely

I-Communic8

Show original language (French)