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SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Occasional Advisor

SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello! Please help configure srtp between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15). My torment has been going on for 2 weeks. SIP TLS registration is successful. The call and conversation without encryption is successful, when srtp is turned on, nothing works. Certificate generated according to the instructions: https://community.polycom.com/t5/VoIP-SIP-Phones/FAQ-How-can-I-setup-a-TLS-connection-for-SIP-signal... .I really count on your help, thank you very much! Сonfiguration and log files are attached.

15 REPLIES 15
Highlighted
Polycom Employee & Community Manager

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello @Grigory ,

Your post ended up in the Spam Filter so I moved this here.

 

I see that you are based in Russia and the hardware we sell in Russia does comply with government regulations and therefore does not support SRTP.

Best Regards

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 2 of 16
Highlighted
Occasional Advisor

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Thank you for deleting my message from spam!

 

Why not? The WEB menu has an option to enable SRTP. And config has SRTP too.

 

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Application SIP Helford6 5.9.6.2327 19-Mar-20 17:55 -->
<!-- Created 11-05-2020 11:20 -->
<!-- Base profile Generic -->
<PHONE_CONFIG>
<!-- Note: The following parameters have been excluded from the export:
reg.1.auth.password=""
-->
<ALL
call.autoAnswer.SIP="1"
call.shared.disableDivert="0"
lcl.datetime.time.24HourClock="1"
lcl.ml.lang="Russian_Russia"
log.level.change.sip="0"
log.level.change.sipp="0"
log.level.change.sipt="0"
log.level.change.srtp="0"
log.level.change.tls="0"
sec.srtp.offer="1"
sec.srtp.require="1"
sec.TLS.LDAP.strictCertCommonNameValidation="0"
sec.TLS.profileSelection.SIP="PlatformProfile2"
sec.TLS.protocol.sip="TLSv1_1"
sec.TLS.SIP.strictCertCommonNameValidation="0"
tcpIpApp.sntp.address="192.168.0.1"
tcpIpApp.sntp.daylightSavings.enable="0"
tcpIpApp.sntp.gmtOffset="0"
tcpIpApp.sntp.gmtOffsetcityID="71"
voIpProt.SIP.outboundProxy.address="192.168.0.90"
voIpProt.SIP.outboundProxy.port="5061"
voIpProt.SIP.outboundProxy.transport="TLS"
reg.1.address="101@192.168.0.90"
reg.1.auth.userId="101"
reg.1.displayName="101"
reg.1.label="101"
reg.1.outboundProxy.address="192.168.0.90"
reg.1.outboundProxy.port="5061"
reg.1.outboundProxy.transport="TLS"
reg.1.serverAutoDiscovery="0"
reg.1.srtp.offer="1"
reg.1.srtp.require="1"
reg.1.type="shared"
sec.TLS.customCaCert.1="-----BEGIN RSA PRIVATE KEY-----
...
-----END RSA PRIVATE KEY-----
-----BEGIN CERTIFICATE-----
...
-----END CERTIFICATE-----"
sec.TLS.customDeviceCert.1="-----BEGIN CERTIFICATE-----
...
-----END CERTIFICATE-----"
sec.TLS.customDeviceKey.1="-----BEGIN PRIVATE KEY-----
...
-----END PRIVATE KEY-----"
sec.TLS.profile.1.caCert.application1="0"
sec.TLS.profile.1.caCert.application2="0"
sec.TLS.profile.1.caCert.application3="0"
sec.TLS.profile.1.caCert.application4="0"
sec.TLS.profile.1.caCert.application6="0"
sec.TLS.profile.1.caCert.application7="0"
sec.TLS.profile.1.caCert.defaultList="0"
sec.TLS.profile.1.caCert.platform1="0"
sec.TLS.profile.1.caCert.platform2="0"
sec.TLS.profile.1.deviceCert="Platform2"
voIpProt.server.1.address="192.168.0.90"
voIpProt.server.1.port="5061"
voIpProt.server.1.transport="TLS"
reg.1.server.1.address="192.168.0.90"
reg.1.server.1.port="5061"
reg.1.server.1.transport="TLS"
/>
</PHONE_CONFIG>
Message 3 of 16
Highlighted
Polycom Employee & Community Manager

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello @Grigory ,

 

simply to my knowledge as the Russian government requires us to.

 

A simple SIP trace at debug should show if a Crypto is offered or not.

 

Please utilize the quoted FAQ to troubleshoot this.

 

If you are still convinced that this should work (as I know it will not for a Russian SKU) please open a support ticket.


In order to raise a support ticket, you need to work with your Poly reseller as they may need to do this for you.

End Customers are usually unable to open a ticket directly with Poly support. Available End User Poly services offerings are detailed here

If this is some sort of an Internet discounter providing your MAC address or your Poly devices serial will enable us to look up who would be able to support you. This may not be who you purchased the Poly device from.

If the unit is no longer within the warranty please be prepared to Pay Per Incident / PPI. This is all outlined in detail here


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 4 of 16
Highlighted
Occasional Advisor

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello! My phone MAC: 0004F280D6F0

I think that if SIP TLS works, then SRTP should work too.

 

This is the Polycom log file when an incoming call is made with SRTP:

 

0511110440|sip |1|00|CStkDialog::TimeOut500ms: Server State txn finished OPTIONS (0x1507378)
0511110440|sip |3|00|CUser::TimeOut500ms: Destroying NoCall object 'Unknown' of type 'CCallNoCall' (0x14eb848)
0511110440|sip |3|00|CCallNoCall::NewCallState 'Unknown'->'Idle' (0x14eb848)
0511110444|copy |4|00|DNS lookup failed for ztp.polycom.com
0511110444|cfg |4|00|Web|[cfgSaProcessRequestC] Failed to download language file from provisioning server, request path Website_dictionary_language_ru-ru.xml
0511110444|utilm|4|00|uBLFCompressed: File /ffs0/languages/Website_dictionary_language_ru-ru.xml.zzz does not exist or is empty
0511110444|cfg |4|00|Web|[cfgSaProcessRequestC] Website_dictionary_language_ru-ru.xml Language file doesn't exist in cache
0511110508|sip |1|00|MsgSipTcpPacket
0511110508|sip |3|00|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0511110508|sip |3|00|CStkDialog::SetAddressLocal localTag set to ''
0511110508|sip |3|00|CStkDialog::SetAddressLocal new address added of 1
0511110508|sip |2|00|CStkDialog::CStkDialog SetAddressLocal from pRequest To: '101' <101@192.168.0.183:0>
0511110508|sip |2|00|CStkDialog::CStkDialog SetAddressLocal Config '101' <101@192.168.0.90:0>
0511110508|sip |2|00|CStkDialog::CStkDialog TAG 'BE90C44-7D95BAA1' generated
0511110508|sip |2|00|CStkDialog::CStkDialog local addr '101' <101@192.168.0.183:0> Tag 'BE90C44-7D95BAA1'
0511110508|sip |2|00|CStkDialog::CStkDialog exit 0x1507378 local list size 1
0511110508|sip |2|00|CLocalIPManager::GetLocalIP Local IP identified is 192.168.0.183 for Remote address 192.168.0.183
0511110508|sip |2|00|CCallBase::IsChallenged COPIED Dialog Tag to pRequest 'BE90C44-7D95BAA1'
0511110508|sip |2|00|CCallBase::IsChallenged 'OPTIONS' Dialog Tag 'BE90C44-7D95BAA1' pRequest Tag 'BE90C44-7D95BAA1' state 'Trying'
0511110508|sip |2|00|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x40ecdfa8)
0511110508|sip |3|00|UA Server Non-INVITE OPTIONS trans state 'callingTrying'->'completed' by 200 resp 65 timeout(0x40ecdfa8)
0511110508|sip |1|00|CStkDialog::SetDialogState: Dialog 'id0391b68b' State 'Trying'->'Confirmed'
0511110508|sip |1|00|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '192.168.0.90' port 5061 returned 1 results
0511110508|sip |1|00|doDnsListLookup(tls): result 0 '192.168.0.90' port 5061 isInBound 0
0511110508|sip |1|00|CTrans:: SendCommand | this=0x40ecdfa8, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0511110508|sip |1|00|Send: (TLS) entry for address 192.168.0.90 port 5061 can Connect 1 canFailOver 1
0511110508|sip |1|00|Send: (TLS) address 192.168.0.90 port 5061 can Connect 1
0511110508|sip |1|00|CPlcmSipTcpSocket::Send TLS queuedTxData = 0 TotalLen 745 loop count 1 maxQueueDepth 40000
0511110508|sip |1|00|CPlcmSipTcpSocket::Send TLS Sent 745 loop count 1
0511110508|sip |2|00|CTrans::InitRetrans for UA Server Non-INVITE OPTIONS state 'completed' Server 1 of 1 (0x40ecdfa8)
0511110527|sip |1|00|MsgSipTcpPacket
0511110527|sip |5|00|OnRecvData : Can not decode the packet
0511110527|sip |5|00|OnRecvData : Can not decode the packet
0511110540|sip |1|00|CStkDialog::TimeOut500ms: Server State txn finished OPTIONS (0x1507378)
0511110540|sip |3|00|CUser::TimeOut500ms: Destroying NoCall object 'Unknown' of type 'CCallNoCall' (0x14eb848)
0511110540|sip |3|00|CCallNoCall::NewCallState 'Unknown'->'Idle' (0x14eb848)
0511110608|sip |1|00|MsgSipTcpPacket
0511110608|sip |3|00|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0511110608|sip |3|00|CStkDialog::SetAddressLocal localTag set to ''
0511110608|sip |3|00|CStkDialog::SetAddressLocal new address added of 1
0511110608|sip |2|00|CStkDialog::CStkDialog SetAddressLocal from pRequest To: '101' <101@192.168.0.183:0>
0511110608|sip |2|00|CStkDialog::CStkDialog SetAddressLocal Config '101' <101@192.168.0.90:0>
0511110608|sip |2|00|CStkDialog::CStkDialog TAG 'BA8516CE-F79E7DCB' generated
0511110608|sip |2|00|CStkDialog::CStkDialog local addr '101' <101@192.168.0.183:0> Tag 'BA8516CE-F79E7DCB'
0511110608|sip |2|00|CStkDialog::CStkDialog exit 0x1507378 local list size 1
0511110608|sip |2|00|CLocalIPManager::GetLocalIP Local IP identified is 192.168.0.183 for Remote address 192.168.0.183
0511110608|sip |2|00|CCallBase::IsChallenged COPIED Dialog Tag to pRequest 'BA8516CE-F79E7DCB'
0511110608|sip |2|00|CCallBase::IsChallenged 'OPTIONS' Dialog Tag 'BA8516CE-F79E7DCB' pRequest Tag 'BA8516CE-F79E7DCB' state 'Trying'
0511110608|sip |2|00|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x40ecdfa8)
0511110608|sip |3|00|UA Server Non-INVITE OPTIONS trans state 'callingTrying'->'completed' by 200 resp 65 timeout(0x40ecdfa8)
0511110608|sip |1|00|CStkDialog::SetDialogState: Dialog 'id025bb34c' State 'Trying'->'Confirmed'
0511110608|sip |1|00|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '192.168.0.90' port 5061 returned 1 results
0511110608|sip |1|00|doDnsListLookup(tls): result 0 '192.168.0.90' port 5061 isInBound 0
0511110608|sip |1|00|CTrans:: SendCommand | this=0x40ecdfa8, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0511110608|sip |1|00|Send: (TLS) entry for address 192.168.0.90 port 5061 can Connect 1 canFailOver 1
0511110608|sip |1|00|Send: (TLS) address 192.168.0.90 port 5061 can Connect 1
0511110608|sip |1|00|CPlcmSipTcpSocket::Send TLS queuedTxData = 0 TotalLen 746 loop count 1 maxQueueDepth 40000
0511110608|sip |1|00|CPlcmSipTcpSocket::Send TLS Sent 746 loop count 1
0511110608|sip |2|00|CTrans::InitRetrans for UA Server Non-INVITE OPTIONS state 'completed' Server 1 of 1 (0x40ecdfa8)
0511110640|sip |1|00|CStkDialog::TimeOut500ms: Server State txn finished OPTIONS (0x1507378)
0511110640|sip |3|00|CUser::TimeOut500ms: Destroying NoCall object 'Unknown' of type 'CCallNoCall' (0x14eb848)
0511110640|sip |3|00|CCallNoCall::NewCallState 'Unknown'->'Idle' (0x14eb848)
0511110657|sip |1|00|MsgSipTcpPacket
0511110657|sip |5|00|OnRecvData : Can not decode the packet
0511110657|sip |5|00|OnRecvData : Can not decode the packet
0511110708|sip |1|00|MsgSipTcpPacket
0511110708|sip |3|00|CStkDialog::CreateRouteSet: transport set to Target URI 'TLS'
0511110708|sip |3|00|CStkDialog::SetAddressLocal localTag set to ''
0511110708|sip |3|00|CStkDialog::SetAddressLocal new address added of 1
0511110708|sip |2|00|CStkDialog::CStkDialog SetAddressLocal from pRequest To: '101' <101@192.168.0.183:0>
0511110708|sip |2|00|CStkDialog::CStkDialog SetAddressLocal Config '101' <101@192.168.0.90:0>
0511110708|sip |2|00|CStkDialog::CStkDialog TAG '58071C98-D7EF035' generated
0511110708|sip |2|00|CStkDialog::CStkDialog local addr '101' <101@192.168.0.183:0> Tag '58071C98-D7EF035'
0511110708|sip |2|00|CStkDialog::CStkDialog exit 0x1507378 local list size 1
0511110708|sip |2|00|CLocalIPManager::GetLocalIP Local IP identified is 192.168.0.183 for Remote address 192.168.0.183
0511110708|sip |2|00|CCallBase::IsChallenged COPIED Dialog Tag to pRequest '58071C98-D7EF035'
0511110708|sip |2|00|CCallBase::IsChallenged 'OPTIONS' Dialog Tag '58071C98-D7EF035' pRequest Tag '58071C98-D7EF035' state 'Trying'
0511110708|sip |2|00|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x40ecdfa8)
0511110708|sip |3|00|UA Server Non-INVITE OPTIONS trans state 'callingTrying'->'completed' by 200 resp 65 timeout(0x40ecdfa8)
0511110708|sip |1|00|CStkDialog::SetDialogState: Dialog 'id02f08f67' State 'Trying'->'Confirmed'
0511110708|sip |1|00|doDnsListLookup(tls): doDnsSrvLookupForARecordList for '192.168.0.90' port 5061 returned 1 results
0511110708|sip |1|00|doDnsListLookup(tls): result 0 '192.168.0.90' port 5061 isInBound 0
0511110708|sip |1|00|CTrans:: SendCommand | this=0x40ecdfa8, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0511110708|sip |1|00|Send: (TLS) entry for address 192.168.0.90 port 5061 can Connect 1 canFailOver 1
0511110708|sip |1|00|Send: (TLS) address 192.168.0.90 port 5061 can Connect 1
0511110708|sip |1|00|CPlcmSipTcpSocket::Send TLS queuedTxData = 0 TotalLen 745 loop count 1 maxQueueDepth 40000
0511110708|sip |1|00|CPlcmSipTcpSocket::Send TLS Sent 745 loop count 1
0511110708|sip |2|00|CTrans::InitRetrans for UA Server Non-INVITE OPTIONS state 'completed' Server 1 of 1 (0x40ecdfa8)
0511110718|copy |4|00|DNS lookup failed for ztp.polycom.com
0511110718|cfg |4|00|Web|[cfgSaProcessRequestC] Failed to download language file from provisioning server, request path Website_dictionary_language_ru-ru.xml
0511110718|utilm|4|00|uBLFCompressed: File /ffs0/languages/Website_dictionary_language_ru-ru.xml.zzz does not exist or is empty
0511110718|cfg |4|00|Web|[cfgSaProcessRequestC] Website_dictionary_language_ru-ru.xml Language file doesn't exist in cache
Message 5 of 16
Highlighted
Occasional Advisor

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

This my FreeBPX config:

 

===pjsip.conf===
#include pjsip_custom.conf
#include pjsip.transports.conf
#include pjsip.endpoint.conf
#include pjsip.aor.conf
#include pjsip.auth.conf
#include pjsip.registration.conf
#include pjsip.identify.conf

[global]
type=global
user_agent=FPBX-15.0.16.42(16.6.2)
default_outbound_endpoint=dpma_endpoint
endpoint_identifier_order=ip,username,anonymous,header,auth_username
default_outbound_endpoint=dpma_endpoint
#include pjsip_custom_post.conf

===pjsip.transports.conf===

[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5061
allow_reload=no
tos=cs3
cos=3
local_net=192.168.0.0/24

[0.0.0.0-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5060
allow_reload=no
tos=cs3
cos=3
local_net=192.168.0.0/24

[0.0.0.0-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
ca_list_file=/etc/pki/tls/certs/ca-bundle.crt
cert_file=/etc/asterisk/keys/server_ca_new.pem
priv_key_file=/etc/asterisk/keys/server_ca_new.key
method=tlsv1_1
verify_client=no
verify_server=no
allow_reload=no
tos=cs3
cos=3
local_net=192.168.0.0/24

===pjsip.endpoint.conf===

[102]
type=endpoint
aors=102
auth=102-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722,h264,mpeg4
context=from-internal
callerid=102 <102>

dtmf_mode=auto
direct_media=no
aggregate_mwi=no
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=yes
trust_id_inbound=yes
media_encryption=sdes
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
send_pai=yes
rtp_symmetric=no
rewrite_contact=yes
force_rport=no
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

[101]
type=endpoint
aors=101
auth=101-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722,h264,mpeg4
context=from-internal
callerid=101 <101>

dtmf_mode=auto
direct_media=no
transport=0.0.0.0-tls
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=yes
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=sdes
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=no
refer_blind_progress=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord

[anonymous]
type=endpoint
context=from-sip-external
allow=all

[dpma_endpoint]
type=endpoint
context=dpma-invalid
context=dpma-invalid

===pjsip.aor.conf===

[102]
type=aor
max_contacts=1
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60

[101]
type=aor
max_contacts=1
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60

===pjsip.auth.conf===

[102-auth]
type=auth
auth_type=userpass
password=12345678
username=102

[101-auth]
type=auth
auth_type=userpass
password=12345678
username=101
Message 6 of 16
Highlighted
Occasional Advisor

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

This was answered by the support service:

 

Hello

Thank you for contacting Poly Global Services.

I understand that you are having an issue with your VVX 400

Support for VoIP Products, SoundStation IP and SoundPoint IP, is extended to Certified VoIP Partners only.

If you are unsure of who your service provider is or if they are unable to assist, we show this unit originally being sold through this Certified Partner.

In addition, the following support options are available to you:

A) A community based support forum can be found at: http://community.polycom.com.

B) Poly’s knowledgebase and frequently asked question can be found at: http://support.polycom.com/kb

C) The product support page can be found at:
https://support.polycom.com/content/support/north-america/usa/en/support/voice.html

Thank you.
Poly Global Services.

Message 7 of 16
Highlighted
Polycom Employee & Community Manager

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello @Grigory ,

 

Your Mac shows as a VX 400,POE,RU (P/N: 2200-46157-114) aka Russia.

 

Poly can sell phones in Russia that can:

 

  • Encrypt the SIP Traffic using TLS
  • disable SRTP in the factory as the license we hold for Russia does not allow us to encrypt the Media Stream

The above explains why you can use TLS but not SRTP. This is the same if the phone would be used for Skype for Business.

 

The above is a requirement by the Russian Government.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 8 of 16
Highlighted
Polycom Employee & Community Manager

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Hello @Grigory ,

 

just to add to your latest comment about the reply received from our support team. I had already outlined the following:

 

  • In order to raise a support ticket, you need to work with your Poly reseller as they may need to do this for you.

    End Customers are usually unable to open a ticket directly with Poly support. Available End User Poly services offerings are detailed here

 

Checking the supplied Mac shows the phone was sold 29/06/2013 via OCS Distribution Deutschland AG.

 

Therefore any support we can officially provide would be via PPI.

 

As I made it clear why this is not working please accept this as a solution.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

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The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. My official "day" Job is 3rd Level support at Poly but I am unable to provide official support via the community.

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Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 9 of 16
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Occasional Advisor

Re: SRTP between Polycom VVX 400 (version 5.9.6.2327) and FreePBX (version 15)

Thank you very much for your comprehensive answer!

Message 10 of 16