Product Bulletin: ScopServ releases chan_pjsip to replace deprecated chan_sip

Product Bulletin: ScopServ releases chan_pjsip to replace deprecated chan_sip

Product Bulletin: ScopServ releases chan_pjsip to replace deprecated chan_sip

As per Sangoma's official product announcement chan_sip is officially deprecated. ScopServ's officially supported release of Asterisk is 18.x so this effects all ScopTEL systems not updated to chan_pjsip.


ScopTEL version pre-requisites for chan_pjsip support

Minimum version scopserv-telephony25-7.31.0.0.20221106-1.el7.scopserv or if you are in a php7 environment the minimum version is scopserv-telephony25-7.31.0.0.20221106-2.el7.php7


New Features and Caveats of chan_pjsip

  1. Feature: Shared Device improvements no longer require pseudo username suffixes thanks to multiple AOR support in chan_pjsip
    1. Example extension 8010 device 2/3 no longer requires username 8010_[23]
      1. chan_sip Shared Device Username Example: 

      2. chan_pjsip example: 
      3. Caveats: This means that any Shared Device softphone or provisioned desk phone must be re-provisioned with the primary username instead of <Extension>_X
      4. It is highly advised to schedule the conversion of chan_sip to chan_pjsip with the re-provisioning of any Shared Device Extensions to avoid end user downtime.
        1. ScopCOMM apps must be reset and new QR codes provisioned. For this reason it is advised to put the ScopCOMM push servers 75.98.128.37 75.98.128.38 into the Telephony Security Whitelist to prevent NAT'd IP's from being blacklisted during re-registrations.
        2. This also applies to any ScopTEAMS registrations. The system admin must edit all User Registrations in the their ScopTEAMS Admin portal and remove the _X suffix from all user accounts and SYNC their accounts. So your ScopTEAMS SBC addresses should also be added to the Security whitelist.
        3. Any phone provisioned in the ScopTEL APS will automatically have its Shared Device authentication <extension>_X changed to use the primary login/authentication defined on the Extension|Authentication tab after a Telephony Commit. So admin must do a Telephony Commit followed by an APS Commit and then reboot the phone so it downloads the correct authentication credentials.

      5. To remove the <username>_X suffix from ScopCOMM Mobile applications already provisioned the QR Code must be resent after the Telephony Commit.
        1. To resend the QR codes to all ScopCOMM Mobile Extensions

        2. To reset the ScopCOMM Mobile app settings and allow a new QR code to be scanned. Open the ScopCOMM App and tap on the 3 dots to enter the configuration menu.

          1. Tap on Reset Application to remove the current configuration since it's no longer valid.

          2. The user must check their email for the new QR Code and scan the code into the ScopCOMM app.
          3. The ScopCOMM app should automatically register with the server using the updated registration credentials.
  2. Feature/Caveat: Any Hotdesk extensions or Agent members will also ring registered Shared Devices concurrently when the primary extension is called.
  3. If ScopCOMM is sharing an extension with another device then both extensions must share the same transport method. Example: If the primary device is a desk phone and configured to use UDP then ScopCOMM must also use the same transport type.
    1. Caveat: Push notifications to ScopCOMM only work if the ScopCOMM transport is configured to use TCP or TLS transports. If ScopCOMM is configured to use the UDP transport then push notifications will fail. If the Extension's transport is configured to use Automatic transport then the transport defaults to UDP.
  4. SRTP/TLS support is greatly improved. Therefore UDP is not a recommended transport when using ScopCOMM! Use either TCP or configure TLS/SRTP.
  5. Load improvements, concurrent channels should improve.
  6. T.38 Fax transmissions will no longer use chan_sip so fax performance should improve.
  7. SIP VoIP Interfaces will use Identify and other parameters to check INVITE's for the proper Class of Service lookup. This fixes a known issue where multiple friend registrations to the same IP address/registrar fail to pass the Incoming Class of Service lookup when the incoming INVITE is parsed. This means it is no longer necessary to use the Multiple Trunk feature on Incoming Lines.
  8. Caveat: It is possible to toggle a server between chan_sip and chan_pjsip but this is a global change and effects all tenants. Only one channel driver can be selected per server and the recommendation is chan_pjsip.
  9. Caveat: Asterisk sip CLI commands are replaced with pjsip commands. Example: 'sip show peers' becomes 'pjsip show aors'. This does not affect ScopSTATS reports in any way.
  10. Greatly improved support for WebRTC using wss transport.
  11. DNS SRV with load balancing is supported: https://www.asterisk.org/pjsip-dns-support/
  12. When configuring an IP address or FQDN for a PJSIP VoIP Interface you must append the :<port_number> as in this example:

How to Switch Channel Drivers from chan_sip to chan_pjsip

Edit Telephony|Configuration|Telephony Modules|Protocols|Module Version and select PJSIP from the drop list.
Save
Commit




Restart Services to load to chan_pjsip

Restart Telephony Server, FastAGI Server and Live Monitoring daemons (Realtime) to Finish changing the channel driver to chan_pjsip (this can be done with root/admin GUI access or by SSH). Or simply reboot the server after the Telephony Dial Plan and APS Commits.



Tools|Telephony|Reset Database to clear astdb registry and Complete the Migration (Mandatory).


This step is mandatory in order to clear chan_sip settings from memory.
You must restart Asterisk and do a Full Commit on Telephony. Phone registrations may take up to 3600 seconds to renew based on the configured registration re-registration interval, which is typically 3600 seconds. To speed this process up you can reboot any SIP phones or softphones so they will not have to wait for the registration interval to expire.
Once this step is completed you can verify chan_pjsip is loaded with asterisk -vr and 'module' commands

*CLI> module show like res_pjsip.so
Module                         Description                              Use Count  Status      Support Level
res_pjsip.so                   Basic SIP resource                       48         Running              core
1 modules loaded



To revert to chan_sip

Edit Telephony|Configuration|Telephony Modules|Protocols|Module Version and select SIP from the drop list by referring to the previous example.
Save
Commit

Restart Services to revert channel driver to chan_sip

Restart Telephony Server, FastAGI Server and Live Monitoring daemons (Realtime) to Finish changing the channel driver to chan_pjsip (this can be done with root/admin GUI access or by SSH). Or simply reboot the server after the Telephony Dial Plan and APS Commits.


P-Asserted Migration pre-requisites for STIR/SHAKEN/ATIS support.

Servers configured with RPID caller ID support should be migrated to P-Asserted because RPID cannot support Caller ID enhancements necessary to implement STIR/SHAKEN/ATIS protocols. P-Asserted will also support Caller ID updates on transferred calls. STIR/SHAKEN/ATIS is a new module provided by chan_pjsip


Migrating to P-Asserted


Use the Extensions|Mass Operations|Caller ID tool to mass edit extensions from RPID to P-Asserted.




Use the Incoming Lines|Mass Operations|Caller ID Options tool to Override CallerID info with P-Asserted-ID (SIP) header? [x]




Change each VoIP Interface to send P-Asserted.



Commit

Polycom, Snom, Aastra/Mitel/Grandstream phones by default use PAI
Yealink phones configured in the ScopTEL APS may require a change to each MAC/Line to enable PAI-FROM.
Verify each line assignment is set to PAI-FROM for all MAC addresses.

Example: 

Once each MAC address/Line is correctly configured Commit APS changes and reboot all Yealink phones to acquire the new setting.

The Following Cisco Phones are not Supported by PJSIP


Model

Lines

image_version

File Type

Notes

3905

1

 CP3905.9-4-1SR2-2

SEPMAC.cnf.xml

 

3911

 

 

 

Not Supported

3951

 

 

 

Not Supported

6901

1

 SIP6901.9-3-1-SR2-3

SEPMAC.cnf.xml

 

6911

1

 SIP6911.9-3-1-SR2-4

SEPMAC.cnf.xml

 

6921

2

 SIP69xx.9-4-1-3SR2

SEPMAC.cnf.xml

 

6941

4

 SIP69xx.9-4-1-3SR2

SEPMAC.cnf.xml

 

6945

4

 SIP69xx.9-4-1-3SR2

SEPMAC.cnf.xml

 

6961

12

 SIP69xx.9-4-1-3SR2

SEPMAC.cnf.xml

 

7811

1

sip78xx.11-5-1-18

SEPMAC.cnf.xml

 

7821

2

sip78xx.11-5-1-18

SEPMAC.cnf.xml

 

7841

4

sip78xx.11-5-1-18

SEPMAC.cnf.xml

 

7861

16

sip78xx.11-5-1-18

SEPMAC.cnf.xml

 

7905

 

CP7905080000SIP060111A

SIPMAC.cnf

End Of Life

7906

1

 SIP11.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7911

1

 SIP11.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7912

1

 CP7912010301SIP050608A

SIPMAC.cnf

End Of Life

7931

24

 SIP31.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7940

2

 P0S3-07-5-00

SIPMAC.cnf

No Call Manager Support

7941

2

 SIP41.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7961

6

 SIP41.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7942

2

 SIP42.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7945

2

 SIP45.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7960

2

 P0S3-07-5-00

SIPMAC.cnf

No Call Manager Support

7962

6

 SIP42.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7965

6

 SIP45.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7970

8

 SIP70.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7971

8

 SIP70.9-4-2SR2-2S

SEPMAC.cnf.xml

 

7975

8

 SIP75.9-4-2SR2-2S

SEPMAC.cnf.xml

 

88XX

5

sip88xx.11-5-1-18

SEPMAC.cnf.xml


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