WarmTransfer
cdr analytics · published

Carrier escalation evidence packages

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

Disputed: 1 source below carries conflicting evidence.

In SIP, the Call-ID header field carries a globally unique identifier for the call and groups the messages belonging to it 48. A SIP dialog is identified by the combination of the Call-ID, the local tag, and the remote tag 49.

Call identifiers across intermediaries

A back-to-back user agent (B2BUA) receives a request as a UAS and generates its own requests as a UAC, maintaining dialog state unlike a proxy 2. Intermediaries such as session border controllers (SBCs) alter call identifiers in signaling, meaning the SIP Call-ID is not suitable for tracking a session end to end 8. To address this, the SIP Session-ID header defined in RFC 7989 allows endpoints, intermediaries, and management systems to identify a session end to end, including across redirection 47. RFC 7989 obsoletes RFC 7329, the earlier pre-standard Session-ID specification 41. A Session-ID value consists of a local UUID with an optional remote parameter carrying the peer UUID, each formatted as 32 lowercase hexadecimal characters, using an all-zero nil UUID when the peer is unknown 46.

WarmTransfer's reading of the sources is that because SBCs and B2BUAs change the Call-ID between legs, an escalation to a carrier should quote the identifiers seen on the carrier-facing leg (its Call-ID, numbers, and UTC-or-zoned timestamp) rather than an internal platform identifier such as a Microsoft Teams Correlation ID 54.

Response codes and PSTN mapping

SIP final failure responses fall into 3 classes: 4xx client error, 5xx server error, and 6xx global failure 50. The SIP Reason header carries a protocol value of SIP or Q.850 with a numeric cause parameter and an optional text parameter, enabling a Q.850 cause to be conveyed in SIP signaling 40. The Reason header may appear in any request within a dialog, in any CANCEL request, and in responses whose status code explicitly allows it, with RFC 3326 highlighting its use in BYE and CANCEL 39.

Under RFC 3398, ISUP causes map to SIP responses:

  • ISUP cause 1 (unallocated number) and cause 3 (no route to destination) map to SIP 404 Not Found 16.
  • ISUP cause 17 (user busy) maps to SIP 486 Busy Here 18.
  • ISUP cause 18 (no user responding) maps to SIP 408 Request Timeout, and cause 19 (no answer from user) maps to SIP 480 Temporarily Unavailable 19.
  • ISUP causes 34, 38, 41, 42, and 47 (no circuit, network out of order, temporary failure, switching equipment congestion, resource unavailable) map to SIP 503 Service Unavailable 20.
  • ISUP cause 127 (interworking unspecified) maps to SIP 500 Server Internal Error 17.

WarmTransfer's reading of the sources is that because several distinct ISUP causes collapse to SIP 503 under RFC 3398, an escalation that reports only the SIP status code loses the PSTN-side reason; including any Q.850 cause from a Reason header preserves it 1.

RTCP quality metrics

An RTCP reception report block carries the fraction of RTP packets lost since the previous SR or RR and the cumulative number of packets lost since reception began 42. RTCP interarrival jitter is expressed in RTP timestamp units, not milliseconds 43. RTCP reception reports also carry LSR (the middle 32 bits of the NTP timestamp from the last sender report) and DLSR (the delay since that report in units of 1/65536 second) 44.

Platform evidence sources

Microsoft Teams Direct Routing

The Teams admin center SIP call flow provides a chronological ladder diagram of SIP messages exchanged between the customer SBC and the Microsoft SIP Proxy for a selected Direct Routing call 37. Viewing this SIP call flow requires the Teams Administrator or Teams Telephony Administrator role 36. Microsoft notes that admins should allow up to 30 minutes for Direct Routing SIP call data to reach the Teams admin center, and call records older than 30 days are not available in SIP call flow 38.

In the Direct Routing tab of the Teams PSTN usage report, the Correlation ID is the unique identifier Microsoft instructs admins to cite when contacting Microsoft Support 26. The Shared Correlation ID exists only in the exported Direct Routing CSV, not in the portal, and links 2 or more related calls such as the original call and a transfer leg 35. Direct Routing report timestamps (Invite, Start, Failure, and End time) are shown in UTC, and all dates in the exported CSV are UTC in ISO 8601 format 28. Direct Routing Invite time represents when the initial INVITE was sent to the SBC on an outbound call or received from the SBC by the Direct Routing SIP Proxy on an inbound call 30.

For failed Direct Routing calls, the report lists Final SIP code, Final Microsoft subcode, and Final SIP phrase as failure reasons 29. A Microsoft response code starting with 560 indicates that the final SIP response was generated by the SBC, with the last 3 digits providing that SIP code (for instance, 560403 indicates the SBC sent 403), in which case Microsoft recommends starting with SBC logs and the SBC vendor 24. Response codes not starting with 560 indicate the final SIP response was generated by a Microsoft service 31. This SIP response code guidance applies strictly to Direct Routing PSTN calls and does not apply to Calling Plan or Operator Connect deployments 25. Direct Routing usage data can be exported for up to 150 days from the current date unless country or regional regulations prohibit retaining it that long 27.

The Teams admin center SBC test cases tab can execute Outbound-Inbound, Simultaneous ring, Media escalation, and Consultative transfer tests against a chosen SBC using test user credentials 34.

Cisco Unified Border Element (CUBE)

Cisco's baseline for CUBE debug collection sets debug and log timestamps to datetime with msec, localtime, show-timezone, and year, and enables service sequence-numbers 12. Cisco's debug baseline specifies a 10 MB logging buffer and disables console and monitor logging, warning that active console and monitor logging can harm router performance and debug accuracy 10. For SIP calls traversing CUBE, Cisco recommends debug ccsip messages and debug ccsip error together with debug voip ccapi inout, which typically has minimal CPU impact and provides call state information 11.

Carrier and CPaaS tooling

Bandwidth advises checking status.bandwidth.com for existing incidents under investigation prior to opening a ticket 6. When submitting a call failure ticket, Bandwidth requests the dialed number, calling number, and the time and time zone of the call 7. Bandwidth requests call failure examples no older than 24 hours because older incidents can be difficult for peer carriers to investigate 3. In the Bandwidth App Call Logs page, the Call ID field represents the SIP Call-ID 4. Bandwidth Call Logs return results from the preceding 30 days, with downloads capped at 10,000 records 5.

For SIP call failures, Telnyx requests the SIP Connection name and ID, direction, CLI, CLD, and date and time with time zone, or alternatively a Call ID, CDR ID, or Call UUID from the portal 53. Telnyx NOC can troubleshoot most issues up to 72 hours after occurrence, but requires examples within 48 hours when carrier partner escalation is needed, and within 24 hours for non-reproducible issues 52. The Telnyx portal SIP Call Flow Tool provides a maximum 3-day lookback window 51.

WarmTransfer's reading of the sources is that the carrier-side example windows documented by Bandwidth (24 hours) and Telnyx (24 to 72 hours) are much shorter than customer-side retention such as Microsoft's 30-day SIP call flow, so the carrier window rather than local retention limits how late an escalation can be raised 9.

Twilio Elastic SIP Trunking call logs in Console can be filtered by date, phone numbers, SIP addresses, and call status, and downloaded as CSV 56. For SIP Interface and Elastic SIP Trunking calls, Twilio suggests comparing local SIP logs and pcaps against the Twilio public pcap available on the call resource page in Console 55. Twilio Voice Trace captures every RTP packet and DTMF key press on account calls so Twilio Support can reconstruct audio 60. Twilio stores Voice Trace captures for 10 days 57. Voice Trace captures contain confidential Twilio system details and cannot be shared outside Twilio, preventing customers from using them as an artefact to give to another carrier 59. Twilio notes that customers may need party consent before enabling Voice Trace (including for HIPAA-enabled accounts), and accounts using Pay or PCI redaction do not capture RTP 58.

Regulatory traceback requests

Under 47 CFR 64.1200(n)(1), a voice service provider in the United States must fully respond within 24 hours to a traceback request originating from the FCC, civil or criminal law enforcement, or the industry traceback consortium 14. The Industry Traceback Group (ITG) requests that requesters provide the receiving party's number, call date and time in UTC, purported caller number, associated campaign, and purpose of the request, with an optional recording or transcript 22. The ITG prioritizes traceback requests whose call examples are less than 10 days old because they have the highest completion rate 23. Voice service providers do not need to register with the ITG in advance to cooperate; the ITG utilizes a secure traceback portal and contacts providers through downstream partners and existing contact paths 21.

See also

See also SIP trace capture and ladder analysis.

Applicability

Applies to: IETF SIP, SIP trunking, IETF SIP-ISUP interworking, IETF RTP, Microsoft Teams Phone Direct Routing, Microsoft Teams admin center PSTN usage report, Cisco CUBE, Bandwidth voice support, Bandwidth App Call Logs, Telnyx support, Telnyx Mission Control portal, carrier trouble ticketing, Twilio Elastic SIP Trunking, Twilio Programmable Voice, FCC 47 CFR 64.1200, Industry Traceback Group traceback request intake, and Industry Traceback Group traceback provider guidance. Deployments: on-premises, multi-tenant, and any. Sources checked 2026-09-24. In certain countries, Teams PSTN and Direct Routing reports mask the last 3 or 4 digits of external phone numbers, including 3 digits in Germany and 4 in France 32. Traceback response obligations under 47 CFR 64.1200(n)(1) apply to voice service providers in the United States 14.

What remains uncertain

Sources disagree regarding data freshness for Microsoft Teams Direct Routing call data: the PSTN usage report documentation states that reports typically reflect 24 to 48 hours latency, whereas the SIP call flow documentation states that admins should allow up to 30 minutes for SIP call data to appear 33. Resolving this Microsoft Direct Routing data latency discrepancy is not covered by the sources below.

Tier-1 carrier trouble ticket evidence requirements and escalation matrices for AT&T, Verizon, and Lumen are not covered by the sources below. ITG Policies and Procedures regarding response data fields and CDR retention are not covered by the sources below. The use of STIR/SHAKEN attestation and the SIP Identity header as carrier escalation evidence is not covered by the sources below. Webex Calling call identifiers and CDR fields usable for carrier escalation are not covered by the sources below. The Operator Connect escalation path and evidence division between Microsoft and operators are not covered by the sources below. Which specific SBCs and carriers implement RFC 7989 Session-ID is not covered by the sources below. Twilio help center guidance on audio troubleshooting and Voice Trace enablement workflows is not covered by the sources below.

See also

Referenced by

Sources

  1. 1
    Because several distinct ISUP causes collapse to SIP 503 under RFC 3398, an escalation that reports only the SIP status code loses the PSTN-side reason; including any Q.850 cause from a Reason header preserves it.inferred
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1 read together with RFC 3326 section 2 · Checked 2026-09-24
  2. 2
    A back-to-back user agent receives a request as a UAS and generates its own requests as a UAC, and unlike a proxy it maintains dialog state.
    RFC 3261 — SIP: Session Initiation Protocol · Section 6, definition of Back-to-Back User Agent · Checked 2026-09-24
  3. 3
    Bandwidth asks that call failure examples be no older than 24 hours because older incidents may be hard for peer carriers to investigate.
    Troubleshooting technical issues and service impairments · Section on call failure reporting · Checked 2026-09-24
  4. 4
    In the Bandwidth App Call Logs page the Call ID field is the SIP Call-ID of the call.
    How do I use the Call Logs page? · Field list, Call ID · Checked 2026-09-24
  5. 5
    Bandwidth Call Logs return results from the past 30 days, and downloads are limited to 10,000 records.
    How do I use the Call Logs page? · Search window and export sections · Checked 2026-09-24
  6. 6
    Bandwidth advises checking status.bandwidth.com for incidents already under investigation before opening a ticket.
    Troubleshooting technical issues and service impairments · Pre-ticket steps section · Checked 2026-09-24
  7. 7
    When reporting a call failure, Bandwidth asks for both the dialed number and the calling number plus the time and time zone of the call.
    Troubleshooting technical issues and service impairments · Section on call failure reporting · Checked 2026-09-24
  8. 8
    RFC 7989 states that intermediaries such as session border controllers alter call identifiers in signaling, so the SIP Call-ID is not suitable for tracking a session end to end.
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Section 1, Introduction · Checked 2026-09-24
  9. 9
    The carrier-side example windows documented by Bandwidth (24 hours) and Telnyx (24 to 72 hours) are much shorter than customer-side retention such as Microsoft's 30-day SIP call flow, so the carrier window rather than local retention limits how late an escalation can be raised.inferred
    Troubleshooting technical issues and service impairments · Call failure reporting section, compared with telnyx-help-contacting-support time limits and ms-learn-dr-sip-call-flow Note · Checked 2026-09-24
  10. 10
    Cisco's CUBE debug baseline uses a 10 MB logging buffer and disables console and monitor logging, which it warns can harm router performance and debug accuracy.
    Configure Debug Collection for CUBE and TDM Gateways · Baseline logging configuration section and console logging guidance · Checked 2026-09-24
  11. 11
    For SIP calls through CUBE, Cisco recommends debug ccsip messages and debug ccsip error together with debug voip ccapi inout, which usually has minimal CPU impact and adds call state information.
    Configure Debug Collection for CUBE and TDM Gateways · SIP debugs section · Checked 2026-09-24
  12. 12
    Cisco's baseline for CUBE debug collection sets debug and log timestamps to datetime with msec, localtime, show-timezone and year, and enables service sequence-numbers.
    Configure Debug Collection for CUBE and TDM Gateways · Baseline logging configuration section · Checked 2026-09-24
  13. 13
    Under 64.1200(n)(2), a gateway or originating provider that receives an FCC Enforcement Bureau Notice of Suspected Illegal Traffic must investigate and block or cease accepting the identified and substantially similar traffic on an ongoing basis.
    47 CFR § 64.1200 - Delivery restrictions · Paragraph (n)(2) · Checked 2026-09-24
  14. 14
    Under 47 CFR 64.1200(n)(1) a voice service provider must fully respond within 24 hours to a traceback request from the FCC, civil or criminal law enforcement, or the industry traceback consortium.
    47 CFR § 64.1200 - Delivery restrictions · Paragraph (n)(1) · Checked 2026-09-24
  15. 15
    The 64.1200(n)(1) 24-hour clock runs only in business hours (8 a.m. to 5:30 p.m.) on business days (Monday to Friday excluding Federal legal holidays).
    47 CFR § 64.1200 - Delivery restrictions · Paragraph (n)(1), business day and business hours definitions · Checked 2026-09-24
  16. 16
    RFC 3398 maps ISUP cause 1 (unallocated number) and cause 3 (no route to destination) to SIP 404 Not Found.
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1, ISUP cause to SIP response table · Checked 2026-09-24
  17. 17
    RFC 3398 maps ISUP cause 127 (interworking unspecified) to SIP 500 Server Internal Error.
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1, ISUP cause to SIP response table · Checked 2026-09-24
  18. 18
    RFC 3398 maps ISUP cause 17 (user busy) to SIP 486 Busy Here.
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1, ISUP cause to SIP response table · Checked 2026-09-24
  19. 19
    RFC 3398 maps ISUP cause 18 (no user responding) to SIP 408 Request Timeout and cause 19 (no answer from user) to SIP 480 Temporarily Unavailable.
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1, ISUP cause to SIP response table · Checked 2026-09-24
  20. 20
    RFC 3398 maps ISUP causes 34, 38, 41, 42 and 47 (no circuit, network out of order, temporary failure, switching equipment congestion, resource unavailable) to SIP 503 Service Unavailable.
    RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping · Section 8.2.6.1, ISUP cause to SIP response table · Checked 2026-09-24
  21. 21
    Providers do not need to register with the ITG in advance to cooperate with traceback requests; the ITG uses a secure traceback portal and reaches providers through contacts including their downstream partners.
    For Providers - Industry Traceback Group · For Providers page, registration and portal paragraphs · Checked 2026-09-24
  22. 22
    The Industry Traceback Group asks requesters to supply the receiving party's number, the call date and time in UTC, the purported caller number, the associated campaign and the purpose of the request, with an optional recording or transcript.
    Traceback Requests - Industry Traceback Group · Traceback request guidelines, required information · Checked 2026-09-24
  23. 23
    The ITG prioritises traceback requests whose call examples are less than ten days old because they are most likely to complete.
    Traceback Requests - Industry Traceback Group · Traceback request guidelines, timing note · Checked 2026-09-24
  24. 24
    A Microsoft response code starting with 560 means the final SIP response was generated by the SBC, with the last three digits giving that SIP code (560403 means the SBC sent 403); Microsoft recommends starting with SBC logs and the SBC vendor.
    Microsoft and SIP response codes · Section Direct Routing error codes, first bullet · Checked 2026-09-24
  25. 25
    Microsoft's SIP response code guidance applies to Direct Routing PSTN calls and does not apply to Calling Plan or Operator Connect deployments.
    Microsoft and SIP response codes · Opening Note · Checked 2026-09-24
  26. 26
    In the Direct Routing tab of the Teams PSTN usage report, the Correlation ID is the unique call identifier Microsoft tells admins to use when contacting Microsoft Support.
    Microsoft Teams PSTN usage report · Direct Routing callout table, callout 5, Correlation ID · Checked 2026-09-24
  27. 27
    Direct Routing usage data can be exported for up to 150 days from the current date unless country or region rules prohibit retaining it that long.
    Microsoft Teams PSTN usage report · Section Exported Direct Routing usage report · Checked 2026-09-24
  28. 28
    Direct Routing report timestamps (Invite, Start, Failure and End time) are shown in UTC, and all dates in the exported CSV are UTC in ISO 8601 format.
    Microsoft Teams PSTN usage report · Direct Routing callout table callout 5; section Exporting the reports · Checked 2026-09-24
  29. 29
    For failed Direct Routing calls the report gives Final SIP code, Final Microsoft subcode and Final SIP phrase as the reasons the call failed.
    Microsoft Teams PSTN usage report · Direct Routing callout table callout 5, Failure time (UTC) · Checked 2026-09-24
  30. 30
    Direct Routing Invite time is when the initial INVITE was sent to the SBC on an outbound call or received from the SBC by the Direct Routing SIP Proxy on an inbound call.
    Microsoft Teams PSTN usage report · Direct Routing callout table callout 5, Invite time (UTC) · Checked 2026-09-24
  31. 31
    Microsoft response codes that do not start with 560 indicate that the final SIP response was generated by a Microsoft service.
    Microsoft and SIP response codes · Section Direct Routing error codes, second bullet · Checked 2026-09-24
  32. 32
    In some countries the Teams PSTN and Direct Routing reports mask the last three or four digits of external phone numbers, for example three digits for Germany and four for France.
    Microsoft Teams PSTN usage report · Section Phone number obfuscation, country table · Checked 2026-09-24
  33. 33
    Microsoft documentation gives differing freshness for Direct Routing call data: the PSTN usage report page says reports usually reflect 24 to 48 hours latency, while the SIP call flow page says to allow up to 30 minutes for SIP call data.disputed
    Microsoft Teams PSTN usage report · Direct Routing callout table, callout 2 · Checked 2026-09-24
  34. 34
    The Teams admin center SBC test cases tab can run Outbound-Inbound, Simultaneous ring, Media escalation and Consultative transfer tests against a chosen SBC using test user credentials.
    Diagnose issues that affect Direct Routing · Section Create a test suite, steps 1-5 · Checked 2026-09-24
  35. 35
    The Shared Correlation ID exists only in the exported Direct Routing CSV, not in the portal, and links two or more related calls such as the original call and a transfer leg.
    Microsoft Teams PSTN usage report · Section About Shared Correlation ID · Checked 2026-09-24
  36. 36
    Viewing the Direct Routing SIP call flow requires the Teams Administrator or Teams Telephony Administrator role.
    SIP call flow for Direct Routing in Microsoft Teams · Section Prerequisites · Checked 2026-09-24
  37. 37
    The Teams admin center SIP call flow shows a chronological ladder diagram of SIP messages exchanged between the customer SBC and the Microsoft SIP Proxy for a selected Direct Routing call.
    SIP call flow for Direct Routing in Microsoft Teams · Intro and section Understanding the SIP call flow view · Checked 2026-09-24
  38. 38
    Microsoft says to allow up to 30 minutes for Direct Routing SIP call data to reach the Teams admin center, and call records older than 30 days are not available in SIP call flow.
    SIP call flow for Direct Routing in Microsoft Teams · Section Access SIP call flow, Note · Checked 2026-09-24
  39. 39
    The Reason header may appear in any request within a dialog, in any CANCEL request, and in responses whose status code explicitly allows it; RFC 3326 highlights its use in BYE and CANCEL.
    RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP) · Sections 2 and 3 · Checked 2026-09-24
  40. 40
    The SIP Reason header carries a protocol value of SIP or Q.850 with a numeric cause parameter and an optional text parameter, so a Q.850 cause can be conveyed in SIP signaling.
    RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP) · Section 2, syntax · Checked 2026-09-24
  41. 41
    RFC 7989 obsoletes RFC 7329, the earlier pre-standard Session-ID specification.
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Document header (Obsoletes: 7329) and Abstract · Checked 2026-09-24
  42. 42
    An RTCP reception report block carries the fraction of RTP packets lost since the previous SR or RR and the cumulative number of packets lost since reception began.
    RTP: A Transport Protocol for Real-Time Applications · Section 6.4.1, fields fraction lost and cumulative number of packets lost · Checked 2026-09-24
  43. 43
    RTCP interarrival jitter is expressed in RTP timestamp units, not milliseconds.
    RTP: A Transport Protocol for Real-Time Applications · Section 6.4.1, field interarrival jitter · Checked 2026-09-24
  44. 44
    RTCP reception reports carry LSR (middle 32 bits of the NTP timestamp from the last sender report) and DLSR (delay since that report in units of 1/65536 second).
    RTP: A Transport Protocol for Real-Time Applications · Section 6.4.1, fields LSR and DLSR · Checked 2026-09-24
  45. 45
    RFC 7989 requires B2BUAs to preserve Session-ID values when forwarding messages so identifiers stay consistent across dialog legs.
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Section 7, B2BUA behaviour · Checked 2026-09-24
  46. 46
    A Session-ID value is a local UUID with an optional remote parameter carrying the peer UUID, each written as 32 lowercase hexadecimal characters, with an all-zero nil UUID used when the peer is unknown.
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Section 5, syntax · Checked 2026-09-24
  47. 47
    The SIP Session-ID header defined in RFC 7989 lets endpoints, intermediaries and management systems identify a session end to end, including across redirection.
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Section 1 · Checked 2026-09-24
  48. 48
    In SIP the Call-ID header field carries a globally unique identifier for the call and groups the messages belonging to it.
    RFC 3261 — SIP: Session Initiation Protocol · Sections 8.1.1.4 and 20.8 · Checked 2026-09-24
  49. 49
    A SIP dialog is identified by the combination of the Call-ID, the local tag and the remote tag.
    RFC 3261 — SIP: Session Initiation Protocol · Section 12 · Checked 2026-09-24
  50. 50
    SIP final failure responses fall into three classes: 4xx client error, 5xx server error and 6xx global failure.
    RFC 3261 — SIP: Session Initiation Protocol · Section 7.2 · Checked 2026-09-24
  51. 51
    The Telnyx portal SIP Call Flow Tool searches a maximum three-day lookback window.
    Telnyx Debugging Tools · SIP Call Flow Tool section · Checked 2026-09-24
  52. 52
    Telnyx NOC can troubleshoot most issues up to 72 hours after occurrence, needs examples within 48 hours if a carrier partner escalation is required, and within 24 hours for non-reproducible issues.
    Best Practices for Contacting Support · Section on time limits for examples · Checked 2026-09-24
  53. 53
    For SIP call failures Telnyx asks for the SIP Connection name and ID, direction, CLI, CLD and date and time with time zone, or alternatively a Call ID, CDR ID or Call UUID from the portal.
    Best Practices for Contacting Support · Section on call failures · Checked 2026-09-24
  54. 54
    Because SBCs and B2BUAs change the Call-ID between legs, an escalation to a carrier should quote the identifiers seen on the carrier-facing leg (its Call-ID, numbers and UTC-or-zoned timestamp) rather than an internal platform ID such as a Teams Correlation ID.inferred
    RFC 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks · Section 1, read with bandwidth-help-call-logs-page Call ID field and ms-learn-pstn-usage-report Correlation ID · Checked 2026-09-24
  55. 55
    For SIP Interface and Elastic SIP Trunking calls Twilio suggests comparing local SIP logs and pcaps with the Twilio public pcap available on the call resource page in Console.
    Voice Insights Frequently Asked Questions · FAQ question How can I address the SIP errors? · Checked 2026-09-24
  56. 56
    Twilio Elastic SIP Trunking call logs in Console can be filtered by date, phone numbers, SIP addresses and call status and downloaded as CSV.
    Troubleshooting your Trunk · Section on Elastic SIP Trunking call log · Checked 2026-09-24
  57. 57
    Twilio stores Voice Trace captures for 10 days.
    What is Voice Trace? · Access and storage section · Checked 2026-09-24
  58. 59
    Voice Trace captures contain confidential Twilio system details and cannot be shared outside Twilio, so they are not an artefact a customer can hand to another carrier.
    What is Voice Trace? · Important restrictions section · Checked 2026-09-24
  59. 60
    Twilio Voice Trace, once enabled, captures every RTP packet and DTMF key press on the account's calls so Twilio Support can reconstruct call audio.
    What is Voice Trace? · Sections on what Voice Trace captures and enabling it · Checked 2026-09-24

Documents

tier 1 standards and regulators

47 CFR § 64.1200 - Delivery restrictions

Legal Information Institute, Cornell Law School (reproducing the eCFR) · 2025-08-29 · accessed 2026-09-24

tier 1 standards and regulators

RFC 3261 — SIP: Session Initiation Protocol

IETF / RFC Editor · 2002-06 · accessed 2026-09-04

tier 1 standards and regulators

RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP)

IETF / RFC Editor · 2002-12-01 · 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 7989: End-to-End Session Identification in IP-Based Multimedia Communication Networks

IETF / RFC Editor · 2016-10-01 · accessed 2026-09-24

tier 1 standards and regulators

RTP: A Transport Protocol for Real-Time Applications

IETF / RFC Editor · 2003-07-01 · accessed 2026-09-24

tier 2 current vendor documentation

Best Practices for Contacting Support

Telnyx · accessed 2026-09-24

tier 2 current vendor documentation

Configure Debug Collection for CUBE and TDM Gateways

Cisco Systems · 2026-08-17 · accessed 2026-09-23

tier 2 current vendor documentation

Diagnose issues that affect Direct Routing

Microsoft · 2026-03-14 · accessed 2026-09-23

tier 2 current vendor documentation

For Providers - Industry Traceback Group

Industry Traceback Group (USTelecom) · accessed 2026-09-24

tier 2 current vendor documentation

How do I use the Call Logs page?

Bandwidth · 2025-11-17 · accessed 2026-09-24

tier 2 current vendor documentation

Microsoft and SIP response codes

Microsoft · 2024-11-22 · accessed 2026-09-24

tier 2 current vendor documentation

Microsoft Teams PSTN usage report

Microsoft · 2026-04-26 · accessed 2026-09-24

tier 2 current vendor documentation

SIP call flow for Direct Routing in Microsoft Teams

Microsoft · 2026-03-25 · accessed 2026-09-23

tier 2 current vendor documentation

Telnyx Debugging Tools

Telnyx · accessed 2026-09-24

tier 2 current vendor documentation

Traceback Requests - Industry Traceback Group

Industry Traceback Group (USTelecom) · 2021-08-12 · accessed 2026-09-24

tier 2 current vendor documentation

Troubleshooting technical issues and service impairments

Bandwidth · 2026-06-10 · accessed 2026-09-24

tier 2 current vendor documentation

Troubleshooting your Trunk

Twilio · 2026-05-06 · accessed 2026-09-24

tier 2 current vendor documentation

Voice Insights Frequently Asked Questions

Twilio · accessed 2026-09-24

tier 2 current vendor documentation

What is Voice Trace?

Twilio · 2025-09-25 · accessed 2026-09-24

Cite this page

APA

WarmTransfer. (2026, September 24). Carrier escalation evidence packages. WarmTransfer. https://warmtransfer.net/knowledge/carrier-escalation-evidence

BibTeX

@misc{warmtransfer-carrier-escalation-evidence,
  title  = {Carrier escalation evidence packages},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/carrier-escalation-evidence},
  note   = {Verified 2026-09-24}
}