legacy adjacent · published

PRI and TDM gateway interworking

Verified 2026-09-24 · 60 sources · tier 1–2

Primary Rate Interface (PRI) and time-division multiplexing (TDM) gateway interworking bridges legacy circuit-switched voice facilities into packet-based SIP environments 60​37. Framing and higher-layer call control protocols govern the translation and encapsulation of signalling and bearer traffic across these boundaries 21​57.

Physical framing and signalling standards

TDM circuits rely on standardized synchronous framing and layer 2 and 3 signalling definitions 21​40​42. ITU-T G.704 (10/98), which is the in-force version, defines synchronous frame structures for 1544 kbit/s (T1) and 2048 kbit/s (E1) hierarchical levels 21. ITU-T G.711 defines pulse code modulation (PCM) of voice frequencies, with its base recommendation dated 11/88 and its latest in-force amendment being Amendment 2 (11/09) 22.

ISDN data link layer signalling is governed by ITU-T Q.921, whose in-force components include the 09/97 version and Amendment 1 (06/00) 40. At layer 3, ITU-T Q.931 provides the ISDN user-network interface specification for basic call control, based on the in-force 05/98 version 42. The most recent in-force component of Q.931 is Amendment 2 (12/23), which introduces support for calling line identification authentication 41. Within private corporate networks, ECMA-143 (4th edition, December 2001) specifies QSIG circuit-switched call control at the Q reference point between Private Integrated Services Network Exchanges (PINXs), corresponding to ISO/IEC 11572 15.

Regulatory changes also affect circuit deployments; in the United States, the Federal Communications Commission (FCC) adopted Report and Order FCC 26-19 (WC Docket Nos. 25-208 and 25-209) on 2026-03-26 and released it on 2026-03-27 16. This rule was published at 91 FR 20913 on 2026-04-20 and took effect on 2026-05-20, with certain sections delayed 16.

SIP and ISUP/QSIG translation

Signalling translation between ISDN User Part (ISUP) and SIP follows RFC 3398, a Proposed Standard published in December 2002 that has no Obsoletes or Updated-by entries on the RFC Editor info page as of 2026-09-24 48. In the ISUP-to-SIP direction, RFC 3398 maps specific release cause values to SIP status codes 3​4:

  • Cause 1 (unallocated number) maps to SIP 404 Not Found 3.
  • Cause 17 (user busy) maps to SIP 486 Busy Here 4.
  • Cause 18 (no user responding) maps to SIP 408 Request Timeout 5.
  • Cause 19 (no answer from user) maps to SIP 480 Temporarily Unavailable 6.
  • Cause 27 (destination out of order) maps to SIP 502 Bad Gateway 7.
  • Cause 28 (address incomplete) maps to SIP 484 Address Incomplete 8.
  • Resource-unavailable causes 34, 38, 41, 42, and 47 map to SIP 503 Service Unavailable 45.
  • Any ISUP cause value not listed in the mapping table SHOULD be answered with SIP 500 Server Internal Error 58.

In the reverse direction, RFC 3398 maps SIP 486 Busy Here to ISUP cause 17 (user busy) and SIP 503 Service Unavailable to ISUP cause 41 (temporary failure) 54​55. An unlisted SIP status code SHOULD result in an ISUP REL with cause 127 (interworking unspecified) 59. WarmTransfer's reading of the sources is that a release cause crossing two RFC 3398 gateways is not guaranteed to survive intact, as seen when cause 34 maps to SIP 503 on the ingress gateway and is subsequently translated to cause 41 on the egress gateway 2.

For QSIG interworking, RFC 4497 was published in May 2006 as BCP 117 to specify SIP and QSIG interworking in corporate networks 50. RFC 4497 covers basic call control only, and signalling for QSIG supplementary services is explicitly outside its scope 49.

Signalling encapsulation and overlap handling

RFC 3372 defines SIP for Telephones (SIP-T) as Best Current Practice BCP 63, published in September 2002 56. SIP-T outlines two core gateway functions: encapsulation, which bridges PSTN signalling across SIP networks inside multipart MIME bodies, and translation, which maps ISUP messages and parameters to SIP messages and headers 57. RFC 3204 registers the MIME media types application/ISUP and application/QSIG to carry those signalling objects within SIP bodies 46. The application/ISUP media type requires a version parameter indicating the ISUP variant and permits an optional base parameter 31.

Digit collection differences between TDM and SIP networks are addressed in RFC 3578 52​39. RFC 3578 notes that native SIP endpoints never generate overlap signalling, carrying complete numbers en bloc within the INVITE Request-URI 52. The preferred method in RFC 3578 is for the gateway to convert overlap signalling to en bloc by analyzing digits and timers to send a single INVITE when the number is complete 39. Under RFC 3578, if timer T10 expires, the gateway sends an INVITE containing the digits collected up to that point 38.

Platform integration and hardware architectures

Enterprise platforms integrate TDM trunks into cloud calling architectures through dedicated voice gateways and SBCs 60​12. Cisco documents support for TDM ISDN circuits in Webex Calling via a Cisco IOS XE Local Gateway provisioned with a TDM PSTN trunk 60. For TDM-to-IP flows on an IOS XE Local Gateway, Cisco employs a two-leg routing architecture using internal loop-back dial-peers to permit ICE media optimization on the IP leg 33. In this Local Gateway configuration, loop-back dial-peers use G.711 a-law or mu-law selected to match the companding method of the underlying ISDN circuit 32.

On Cisco ISR 4000 Series routers, network-interface-module (NIM) multiflex trunk (T1/E1 voice) modules support only PVDM4 DSPs, while PVDM2 and PVDM3 DSPs are unsupported 28. Voice applications on the ISR 4000 platform require at least the Unified Communications Technology Package 30. Clocking on the ISR 4000 is configured using network-clock synchronization automatic and sets a T1/E1 controller as an input source using network-clock input-source priority 27.

Cisco established an end-of-sale date of 2023-11-07 and a last ship date of 2024-02-06 for ISR4221, ISR4321, ISR4331, ISR4351, ISR4431, and ISR4451 platforms 24. Software maintenance releases for these ISR 4000 models ended on 2025-08-31, with the last date of support and end of vulnerability and security support scheduled for 2028-11-30 26. IOS XE 17.9.x and 17.12.x represent the final release trains supported on these ISR 4000 platforms 25. Cisco identifies Catalyst 8200 and Catalyst 8300 Series edge platforms as replacement products for the end-of-sale ISR 4000 hardware 29.

Microsoft Teams Direct Routing lists legacy PBXs and analog devices as supported user types, including in environments where Calling Plan or Operator Connect serves other tenant users 12. Microsoft supports Direct Routing exclusively with certified SBCs and may decline support cases involving uncertified devices 9. Certified SBCs handle SIP-to-Teams interworking using supported codecs including SILK, G.711, G.722, G.729, and AMR-WB, with AMR-WB limited to non-bypass mode 10. Direct Routing certifications apply to specific firmware versions, permitting higher versions within the same major.minor release 11. Microsoft is not accepting new nominations for Direct Routing SBC certification until further notice 14.

AudioCodes positions the Mediant 1000 as a hybrid TDM/SIP gateway and SBC supporting up to 6 E1 or 8 T1/J1 trunks and up to 192 concurrent voice sessions 37. The AudioCodes Mediant 1000B SBC is certified for Microsoft Teams Direct Routing with media bypass support on software 7.40A.600 13.

See also

Applicability

Applies to: Cisco Webex Calling Local Gateway, AudioCodes Mediant 1000, ITU-T G.704, IETF RFC 3372, ITU-T Q.921, ITU-T Q.931, ITU-T G.711, Ecma International ECMA-143, FCC 26-19, IETF RFC 3398, IETF RFC 4497, IETF RFC 3204, IETF RFC 3578, Microsoft Teams Phone Direct Routing, and Cisco ISR 4000 Series. Deployments: on-premises, multi-tenant, any, and carrier. Sources checked 2026-09-24. Specific regional applicability applies to the United States for FCC Report and Order FCC 26-19 under WC Docket Nos. 25-208 and 25-209 16. Documented release boundaries apply to Cisco IOS XE release trains 17.9.x and 17.12.x on designated ISR 4000 Series routers, as well as AudioCodes Mediant 1000B software version 7.40A.600 25​13.

What remains uncertain

Carrier-specific ISDN PRI and DS1 discontinuance schedules are not covered by the sources below. Whether FCC 26-19 grandfathering covers business ISDN PRI and which sections have delayed effect is not covered by the sources below. Catalyst 8200 and 8300 support for T1/E1 voice NIMs and PVDM4 as ISR 4000 replacements is not covered by the sources below. TDM gateway support statements for other cloud calling platforms are not covered by the sources below. Vendor default Q.850 cause to SIP mappings on CUBE and AudioCodes and how to override them are not covered by the sources below. T1/E1 clock slip and framing error troubleshooting on voice gateways is not covered by the sources below. NFAS and D-channel backup across multiple PRIs are not covered by the sources below.

See also

Explains

Referenced by

Sources

  1. 1
    RFC 3398 maps ISUP ANM or CON (answer) to SIP 200 OK.
  2. 2
    A release cause that crosses two RFC 3398 gateways is not guaranteed to survive intact; for example cause 34 becomes SIP 503 on the first gateway and SIP 503 becomes cause 41 on the second.inferred
  3. 3
    RFC 3398 maps ISUP release cause 1 (unallocated number) to SIP 404 Not Found.
  4. 4
    RFC 3398 maps ISUP release cause 17 (user busy) to SIP 486 Busy Here.
  5. 5
    RFC 3398 maps ISUP release cause 18 (no user responding) to SIP 408 Request Timeout.
  6. 6
    RFC 3398 maps ISUP release cause 19 (no answer from user) to SIP 480 Temporarily Unavailable.
  7. 7
    RFC 3398 maps ISUP release cause 27 (destination out of order) to SIP 502 Bad Gateway.
  8. 8
    RFC 3398 maps ISUP release cause 28 (address incomplete) to SIP 484 Address Incomplete.
  9. 9
    Microsoft supports Teams Phone with Direct Routing only when a Microsoft-certified SBC is used and may decline support cases that involve non-certified devices.
    Plan Direct Routing · Section 'Support boundaries' · Checked 2026-09-24
  10. 10
    Direct Routing supports SILK, G.711, G.722, G.729 and AMR-WB between Teams and the SBC, with AMR-WB supported in non-bypass mode only.
    Plan Direct Routing · Section 'Supported codecs' · Checked 2026-09-24
  11. 11
    Direct Routing certification is granted to specific SBC firmware versions; higher firmware is supported as long as the major.minor version is the same.
    Session Border Controllers certified for Direct Routing - Microsoft Teams · Note in introductory section · Checked 2026-09-24
  12. 12
    Microsoft's Direct Routing planning page lists analog devices and legacy PBXs as a user type served through Direct Routing, including in tenants that use Calling Plan or Operator Connect for other users.
    Plan Direct Routing · Sections 'Direct Routing with Calling Plan and Operator Connect' and 'Example deployment' · Checked 2026-09-24
  13. 13
    The AudioCodes Mediant 1000B SBC is on Microsoft's certified list for Direct Routing, with media bypass support and software 7.40A.600 listed as supported.
    Session Border Controllers certified for Direct Routing - Microsoft Teams · Table 'Certified SBC vendors', AudioCodes Mediant 1000B SBC row · Checked 2026-09-24
  14. 14
    Microsoft states it is not accepting new nominations for Direct Routing SBC certification until further notice.
    Session Border Controllers certified for Direct Routing - Microsoft Teams · Introductory paragraphs, before 'Certified SBC vendors' · Checked 2026-09-24
  15. 15
    ECMA-143 (4th edition, December 2001) specifies QSIG circuit-switched call control at the Q reference point between PINXs and corresponds to ISO/IEC 11572.
  16. 16
    The FCC adopted the network-modernization Report and Order FCC 26-19 (WC Docket Nos. 25-208 and 25-209) on 2026-03-26 and released it 2026-03-27. It was published at 91 FR 20913 on 2026-04-20 and took effect 2026-05-20, with certain sections delayed.
    Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19) · Federal Register preamble: AGENCY, ACTION, DATES and document header · Checked 2026-09-24
  17. 17
    FCC 26-19 set a uniform 31-day automatic grant period for technology-transition Section 214 discontinuance applications, whether the carrier is dominant or non-dominant.
    Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19) · Report and Order paras. 11 and 48-49 as reported · Checked 2026-09-24
  18. 18
    Under FCC 26-19, a carrier discontinuing services that support interconnection trunks or 911 traffic must designate a coordination contact at least 90 days before filing and coordinate with 911 authorities and service providers.
    Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19) · Report and Order paras. 57-63 as reported · Checked 2026-09-24
  19. 19
    FCC 26-19 gives carriers blanket Section 214(a) authority to grandfather legacy voice services, data services below 25/3 Mbps and copper-provisioned interconnected VoIP without filing an application, after notifying customers.
    Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19) · Report and Order paras. 50-56 as reported · Checked 2026-09-24
  20. 20
    FCC 26-19 eliminated the Commission filing requirements in the network change disclosure rules, while incumbent carriers keep public notice and direct-notice obligations to interconnecting carriers and 911 providers.
    Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19) · SUMMARY; Report and Order paras. 1-10 as reported · Checked 2026-09-24
  21. 21
    ITU-T G.704 (10/98), the in-force version, defines the synchronous frame structures for the 1544 kbit/s (T1) and 2048 kbit/s (E1) hierarchical levels, among others.
  22. 22
    ITU-T G.711 defines pulse code modulation (PCM) of voice frequencies; the base recommendation is dated 11/88 and its latest amendment in force is Amendment 2 (11/09).
    G.711: Pulse code modulation (PCM) of voice frequencies · Landing page title and in-force version list · Checked 2026-09-24
  23. 23
    On ISR 4000 Series digital voice modules the card type t1 or card type e1 command must set the module type before the T1 or E1 controller is configured.
    Configure Integrated Services Routers 4000 Series Digital Voice · Configure section, card type step · Checked 2026-09-24
  24. 24
    Cisco's end-of-sale date for the ISR4221, ISR4321, ISR4331, ISR4351, ISR4431 and ISR4451 was 2023-11-07; the last ship date was 2024-02-06.
    End-of-Sale and End-of-Life Announcement for the Cisco ISR4200, ISR4300 and select ISR4400 Series Platform · Milestones table: End-of-Sale Date and Last Ship Date rows · Checked 2026-09-24
  25. 25
    Cisco states that IOS XE 17.9.x and 17.12.x are the last release trains supported on the listed ISR 4000 platforms.
  26. 26
    The last date of support and the end of vulnerability and security support for those ISR 4000 models is 2028-11-30; software maintenance releases ended 2025-08-31.
    End-of-Sale and End-of-Life Announcement for the Cisco ISR4200, ISR4300 and select ISR4400 Series Platform · Milestones table: End of SW Maintenance, End of Vulnerability/Security Support and Last Date of Support rows · Checked 2026-09-24
  27. 27
    Cisco's ISR 4000 digital voice guide configures clocking with network-clock synchronization automatic and designates a T1/E1 controller as the input source with network-clock input-source priority.
    Configure Integrated Services Routers 4000 Series Digital Voice · Configure section, network clock step · Checked 2026-09-24
  28. 28
    On ISR 4000 Series routers the NIM multiflex trunk (T1/E1 voice) modules use only PVDM4 DSPs; PVDM2 and PVDM3 are not supported on the platform.
    Configure Integrated Services Routers 4000 Series Digital Voice · DSP / PVDM4 section · Checked 2026-09-24
  29. 29
    Cisco names Catalyst 8200 and Catalyst 8300 Series edge platforms as the replacement products for the end-of-sale ISR 4000 models.
    End-of-Sale and End-of-Life Announcement for the Cisco ISR4200, ISR4300 and select ISR4400 Series Platform · Product part numbers table, replacement column · Checked 2026-09-24
  30. 30
    Cisco states that voice applications on ISR 4000 Series routers require at least the Unified Communications Technology Package.
    Configure Integrated Services Routers 4000 Series Digital Voice · Requirements section · Checked 2026-09-24
  31. 31
    The application/ISUP media type requires a version parameter identifying the ISUP variant and allows an optional base parameter.
    RFC 3204: MIME media types for ISUP and QSIG Objects · Section 3, application/ISUP registration · Checked 2026-09-24
  32. 32
    In the Cisco Local Gateway TDM example the loop-back dial-peers use G.711 a-law or mu-law chosen to match the companding method of the ISDN service.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Section 'Configure Local Gateway with a TDM PSTN trunk', loop-back dial-peer step · Checked 2026-09-24
  33. 33
    For TDM-IP call flows on a Webex Calling Local Gateway, Cisco uses a two-leg call routing process with internal loop-back dial-peers so that ICE media optimization can be applied on the IP leg.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Section 'Configure Local Gateway with a TDM PSTN trunk' · Checked 2026-09-24
  34. 34
    Webex Calling Local Gateway requires Cisco IOS XE 17.6.1a or later for registration-based trunks and 17.9.1a or later for certificate-based trunks.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Requirements / prerequisites section · Checked 2026-09-24
  35. 35
    The Cisco Local Gateway TDM section sends readers to the Configuring ISDN PRI documentation for ISDN setup and shows an E1 PRI example that uses pri-group timeslots 1-31.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Section 'Configure Local Gateway with a TDM PSTN trunk' · Checked 2026-09-24
  36. 36
    The Cisco Local Gateway TDM example marks loop-around calls with over-decadic prefixes such as A1A and A2A. Voice translation rules add these tags on ingress and strip them before the call leaves on the destination trunk.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Section 'Configure Local Gateway with a TDM PSTN trunk', translation-rule steps · Checked 2026-09-24
  37. 37
    AudioCodes describes the Mediant 1000 as a hybrid TDM/SIP gateway and SBC supporting up to 6 E1 or 8 T1/J1 trunks and 192 concurrent voice sessions.
    Mediant 1000 Media Gateway · Product overview / key specifications section · Checked 2026-09-24
  38. 38
    Under RFC 3578, when timer T10 expires the gateway sends an INVITE containing the digits collected so far.
  39. 39
    RFC 3578's preferred approach is for the gateway to convert overlap to en bloc, collecting digits by number analysis and timers and sending one INVITE once the number is complete.
  40. 41
    The most recent in-force Q.931 component is Amendment 2 (12/23), which adds support for calling line identification authentication.
    Q.931: ISDN user-network interface layer 3 specification for basic call control · Landing page in-force list, Amendment 2 (12/23) row · Checked 2026-09-24
  41. 42
    ITU-T Q.931 is the ISDN user-network interface layer 3 specification for basic call control; its base in-force version is dated 05/98.
    Q.931: ISDN user-network interface layer 3 specification for basic call control · Landing page title and in-force version list · Checked 2026-09-24
  42. 43
    An RFC 4497 gateway appears as a PINX to the QSIG network and as a SIP endpoint to the IP network.
  43. 44
    Under RFC 4497 a gateway that receives a QSIG SETUP maps it to a SIP INVITE and derives the Request-URI from the QSIG called party number.
  44. 45
    RFC 3398 maps the resource-unavailable ISUP causes 34, 38, 41, 42 and 47 to SIP 503 Service Unavailable.
  45. 46
    RFC 3204 registers the MIME media types application/ISUP and application/QSIG for carrying those signalling objects in SIP bodies.
    RFC 3204: MIME media types for ISUP and QSIG Objects · Section 3 (Proposed media types) · Checked 2026-09-24
  46. 47
    The RFC Editor lists RFC 3204 as updated by RFC 3459 and RFC 5621.
    Information on RFC 3204 · Updated by field · Checked 2026-09-24
  47. 48
    RFC 3398, which maps ISUP to SIP, is a Proposed Standard published in December 2002 and has no Obsoletes or Updated-by entries on the RFC Editor info page as of 2026-09-24.
    Information on RFC 3398 · Status and Updated-by fields · Checked 2026-09-24
  48. 49
    RFC 4497 covers basic call only; signalling for QSIG supplementary services is explicitly outside its scope.
  49. 50
    RFC 4497, published May 2006 as BCP 117, specifies interworking between SIP and QSIG within corporate networks.
    RFC 4497: Interworking between the Session Initiation Protocol (SIP) and QSIG · Document header and Abstract · Checked 2026-09-24
  50. 51
    For SIP-bridging (PSTN to SIP to PSTN transit), RFC 3398 requires the SIP messages to carry the ISUP message as a MIME body so it can be reproduced at the far gateway.
  51. 52
    RFC 3578 states that native SIP endpoints never generate overlap signalling; SIP carries the complete number en bloc in the INVITE Request-URI.
  52. 53
    RFC 3578 allows a gateway to generate SIP overlap signalling (multiple INVITEs) only if the SIP routing infrastructure ensures the INVITEs reach a single gateway.
  53. 54
    In the SIP-to-ISUP direction RFC 3398 maps SIP 486 Busy Here to ISUP cause 17 (user busy).
  54. 55
    In the SIP-to-ISUP direction RFC 3398 maps SIP 503 Service Unavailable to ISUP cause 41 (temporary failure).
  55. 56
    RFC 3372 (SIP-T: Context and Architectures) is Best Current Practice BCP 63, published September 2002.
  56. 57
    SIP-T identifies two gateway functions: encapsulation, which bridges PSTN signalling across a SIP network in multipart MIME bodies, and translation, which maps ISUP messages and parameters to SIP messages and headers.
  57. 58
    Under RFC 3398, an ISUP cause value not listed in the mapping table SHOULD be answered with SIP 500 Server Internal Error.
  58. 59
    Under RFC 3398, a SIP status code not listed in the mapping table SHOULD result in an ISUP REL with cause 127 (interworking unspecified).
  59. 60
    Cisco documents that TDM ISDN circuits are supported with Webex Calling through a Cisco IOS XE Local Gateway with a TDM PSTN trunk.
    Configure Local Gateway on Cisco IOS XE for Webex Calling · Section 'Configure Local Gateway with a TDM PSTN trunk' · Checked 2026-09-24

Documents

tier 1 standards and regulators

G.704: Synchronous frame structures used at 1544, 6312, 2048, 8448 and 44 736 kbit/s hierarchical levels

ITU-T · 1998-10-01 · accessed 2026-09-24

tier 1 standards and regulators

G.711: Pulse code modulation (PCM) of voice frequencies

ITU-T · 1988-11-01 · accessed 2026-09-24

tier 1 standards and regulators

Information on RFC 3204

RFC Editor · 2001-12-01 · accessed 2026-09-24

tier 1 standards and regulators

Information on RFC 3398

RFC Editor · 2002-12-01 · accessed 2026-09-24

tier 1 standards and regulators

Q.921: ISDN user-network interface - Data link layer specification

ITU-T · 2000-06-01 · accessed 2026-09-24

tier 1 standards and regulators

Q.931: ISDN user-network interface layer 3 specification for basic call control

ITU-T · 2023-12-01 · accessed 2026-09-24

tier 1 standards and regulators

Reducing Barriers to Network Improvements and Service Changes (Final rule, FCC 26-19)

Office of the Federal Register via GovInfo · 2026-04-20 · accessed 2026-09-24

tier 1 standards and regulators

RFC 3204: MIME media types for ISUP and QSIG Objects

RFC Editor / IETF · 2001-12-01 · accessed 2026-09-24

tier 1 standards and regulators

RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures

RFC Editor / IETF · 2002-09-01 · accessed 2026-09-24

tier 1 standards and regulators

RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping

IETF / RFC Editor · 2002-12-01 · accessed 2026-09-24

tier 1 standards and regulators

RFC 4497: Interworking between the Session Initiation Protocol (SIP) and QSIG

RFC Editor / IETF · 2006-05-01 · accessed 2026-09-24

tier 2 current vendor documentation

Configure Integrated Services Routers 4000 Series Digital Voice

Cisco Systems · 2023-11-09 · accessed 2026-09-24

tier 2 current vendor documentation

Configure Local Gateway on Cisco IOS XE for Webex Calling

Cisco Systems, Inc. (Webex Help Center) · 2026-07-09 · accessed 2026-09-04

tier 2 current vendor documentation

End-of-Sale and End-of-Life Announcement for the Cisco ISR4200, ISR4300 and select ISR4400 Series Platform

Cisco Systems · 2022-11-07 · accessed 2026-09-24

tier 2 current vendor documentation

Mediant 1000 Media Gateway

AudioCodes · accessed 2026-09-24

tier 2 current vendor documentation

Plan Direct Routing

Microsoft (Microsoft Learn) · 2026-08-21 · accessed 2026-09-06

tier 2 current vendor documentation

Session Border Controllers certified for Direct Routing - Microsoft Teams

Microsoft · 2026-05-27 · accessed 2026-09-14

Cite this page

APA

WarmTransfer. (2026, September 24). PRI and TDM gateway interworking. WarmTransfer. https://warmtransfer.net/knowledge/legacy-pri-tdm

BibTeX

@misc{warmtransfer-legacy-pri-tdm,
  title  = {PRI and TDM gateway interworking},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/legacy-pri-tdm},
  note   = {Verified 2026-09-24}
}