#telephony — Public Fediverse posts
Live and recent posts from across the Fediverse tagged #telephony, aggregated by home.social.
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Anyone has recommendations for nice old #phones? Was considering a #Nokia 5110 so far.
Just want it for #telephony, anything else is an addon to me ^^ -
Anyone has recommendations for nice old #phones? Was considering a #Nokia 5110 so far.
Just want it for #telephony, anything else is an addon to me ^^ -
Anyone has recommendations for nice old #phones? Was considering a #Nokia 5110 so far.
Just want it for #telephony, anything else is an addon to me ^^ -
RE: https://hachyderm.io/@sen/116618477936301567
absolutely fascinating: watch workers physically cut massive telephone exchange cables to disconnect the telephone exchange from the old electromechanical switching service, to a new electronic switching service.
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RE: https://hachyderm.io/@sen/116618477936301567
absolutely fascinating: watch workers physically cut massive telephone exchange cables to disconnect the telephone exchange from the old electromechanical switching service, to a new electronic switching service.
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RE: https://hachyderm.io/@sen/116618477936301567
absolutely fascinating: watch workers physically cut massive telephone exchange cables to disconnect the telephone exchange from the old electromechanical switching service, to a new electronic switching service.
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RE: https://hachyderm.io/@sen/116618477936301567
absolutely fascinating: watch workers physically cut massive telephone exchange cables to disconnect the telephone exchange from the old electromechanical switching service, to a new electronic switching service.
-
RE: https://hachyderm.io/@sen/116618477936301567
absolutely fascinating: watch workers physically cut massive telephone exchange cables to disconnect the telephone exchange from the old electromechanical switching service, to a new electronic switching service.
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For those of you in the USA, or nearby and on the NANP (North American Numbering Plan, with +1 country code), have any of you ever gotten a spam text from a shortcode?
#telephony #phone #voip #telephone #sms #usa #unitedstates #nanp
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For those of you in the USA, or nearby and on the NANP (North American Numbering Plan, with +1 country code), have any of you ever gotten a spam text from a shortcode?
#telephony #phone #voip #telephone #sms #usa #unitedstates #nanp
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For those of you in the USA, or nearby and on the NANP (North American Numbering Plan, with +1 country code), have any of you ever gotten a spam text from a shortcode?
#telephony #phone #voip #telephone #sms #usa #unitedstates #nanp
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For those of you in the USA, or nearby and on the NANP (North American Numbering Plan, with +1 country code), have any of you ever gotten a spam text from a shortcode?
#telephony #phone #voip #telephone #sms #usa #unitedstates #nanp
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For those of you in the USA, or nearby and on the NANP (North American Numbering Plan, with +1 country code), have any of you ever gotten a spam text from a shortcode?
#telephony #phone #voip #telephone #sms #usa #unitedstates #nanp
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Migrating Azure Storage Accounts for AudioCodes SBC Deployment from GPv1 to GPv2
Microsoft has announced the retirement of legacy general-purpose v1 (GPv1) storage accounts, requiring customers to migrate to general-purpose v2 (GPv2) by October 13, 2026.
While this may sound like a simple backend change, some environments still rely on older deployments including virtual machines and network appliances that were originally provisioned years ago.
In my case, I recently revisited an older Azure deployment running an AudioCodes Mediant VE Session Border Controller (SBC) for Microsoft Teams Direct Routing. The SBC was still backed by a legacy GPv1 storage account, making it a perfect candidate to walk through the migration process.
Table of Contents
- Initial Situation
- Important Clarification: Does This Affect the SBC?
- Why You Should Migrate?
- Performing the Upgrade (GPv1 to GPv2)
- Observations from my migration
- Things to Watch Out For
- Conclusion,opinion and summary
- Resources
Initial Situation
The environment was fairly typical for earlier Azure-based SBC deployments:
- Virtual Machine: AudioCodes Mediant VE SBC
- Purpose: Microsoft Teams Direct Routing
- Deployment age: Several years old
- Storage Account Type: GPv1 (legacy)
- VM Disk Type: Managed Disks
At the time of deployment, GPv1 was standard and widely used. However, with Microsoft now retiring GPv1 storage accounts, action is required to avoid future disruptions.
Important Clarification: Does This Affect the SBC?
Before performing the migration, one key question arises:
Will changing the storage account impact the running SBC or the VM itself?
Short answer: No (in most cases)
Here’s why:
- Azure VMs use Managed Disks, which are independent of the storage account type
- The storage account is typically only used for:
- Boot diagnostics
- Logging
- Optional file storage
So, this means:
- The SBC runtime is not directly affected
- There is no need to redeploy the VM
- The migration is generally low-risk
Why You Should Migrate?
Microsoft clearly states that:
- GPv1 accounts will be retired on October 13, 2026
- Existing accounts will be automatically migrated if no action is taken
- GPv2 provides:
- Access to new Azure Storage features
- Improved performance
- Support for zonal redundancy (ZRS)
From a consulting perspective: In my opinion, it’s better to migrate proactively rather than rely on automatic changes.
Performing the Upgrade (GPv1 to GPv2)
The migration itself is straightforward and easy.
Step 1: Open the Storage Account
Navigate to:
- Azure Portal
- Storage Accounts
- Your GPv1 Account
Step 2: Upgrade Configuration
In the Configuration section, you’ll find the option to upgrade the account type.
- Change from: General-purpose v1 to General-purpose v2
Step 3: Confirm the Upgrade
- Review the changes
- Confirm the upgrade
Step 4: Validate the Result
After a few moments:
- Storage account type shows GPv2
- No interruption to the VM or SBC as far as I could see
Observations from my migration
From my test with the AudioCodes SBC:
- ✅ No downtime observed: The SBC continued running without any interruption.
- ✅ No impact on voice services: Active calls and signaling were unaffected.
- ✅ No reconfiguration required: All existing settings (diagnostics, logging) remained intact.
Things to Watch Out For
Even though the migration is simple, there are a few considerations:
- ⚠️ Pricing changes
- GPv2 uses a different pricing model (e.g., access tiers)
- Costs may slightly increase depending on usage
- ⚠️ Access Tier settings
- New options like Hot / Cool / Archive
- Review defaults after migration
- ⚠️ Legacy integrations
- Rare, but older scripts or tools may assume GPv1 behavior
Conclusion,opinion and summary
Migrating from GPv1 to GPv2 is:
- ✅ Simple
- ✅ Low risk
- ✅ Required by Microsoft
For workloads like an AudioCodes Mediant VE SBC on Azure, the impact is minimal because the VM itself is not directly tied to the storage account.
The key takeaway:
Even in legacy deployments, this migration can typically be done without downtime or service impact.
If you are managing older Azure environments especially in Teams Direct Routing scenarios it’s worth reviewing:
- Storage account types
- Public IP SKUs
- Load balancer configurations
Many of these legacy components are now being phased out, and proactive upgrades help avoid last-minute pressure.
Resources
- Legacy Blob Storage account retirement overview – Azure Storage | Microsoft Learn
- How to change a Azure basic to a standard public IP? – erik365.blog
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Microsoft Teams Call Transfer Update
Microsoft Teams plans to roll out an updated call transfer experience for desktop users on Windows and macOS. The update focuses on speeding up transfers and making it easier for users to reach the right contact. There will be two buttons for each action one and new normal will be “blind transfers” by default instead of “safe transfers”.
Source: Microsoft 365 Roadmap ID 561028What’s changing?
- Transfer and Consult transfer become one‑click actions (a button per action)
- Suggested transfer targets
- New default: blind transfer (safe transfer via the “Ring Back” toggle as needed)
Rollout timeline
- Targeted Release: Early June 2026
- General Availability: Late June 2026
There is no action required except informing users and updating internal documentations and training material.
Resources
- Microsoft 365 Roadmap ID 561028
- Microsoft Message Center MC1296872
- Transfer a call in Microsoft Teams – Microsoft Support
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Other than some framing and signalling silliness on the span, we have a working circuit and channel bank now.
https://neon.cybre.stream/w/pd5XGMwHtCvcBtiafw166W
The silliness pervades. :bec_smile:
(caution, video contains loud ringing)
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Other than some framing and signalling silliness on the span, we have a working circuit and channel bank now.
https://neon.cybre.stream/w/pd5XGMwHtCvcBtiafw166W
The silliness pervades. :bec_smile:
(caution, video contains loud ringing)
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Other than some framing and signalling silliness on the span, we have a working circuit and channel bank now.
https://neon.cybre.stream/w/pd5XGMwHtCvcBtiafw166W
The silliness pervades. :bec_smile:
(caution, video contains loud ringing)
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Other than some framing and signalling silliness on the span, we have a working circuit and channel bank now.
https://neon.cybre.stream/w/pd5XGMwHtCvcBtiafw166W
The silliness pervades. :bec_smile:
(caution, video contains loud ringing)
-
Other than some framing and signalling silliness on the span, we have a working circuit and channel bank now.
https://neon.cybre.stream/w/pd5XGMwHtCvcBtiafw166W
The silliness pervades. :bec_smile:
(caution, video contains loud ringing)
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Saw this on another site, looks like something I might try:
https://wandel.ca/homepage/pbx.html
Cc: @lpbkdotnet
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When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
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When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
-
When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
-
When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
-
When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
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This looks rather interesting!
https://interfacecraft.online/blog/2026/desktop-phone-connected-to-freebsd-server/
Not sure if you've seen this, @lpbkdotnet -- but I thought of you...
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The advantage of rotary dialing was the reduction of drunk dialing and spam calls.
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Anyone familiar with old SIP equipment? Specifically an Audiocodes MP-1xx FXS?
I have a two port 112 unit that I'm trying to get port 1 to call port 2 without having a PBX or a SIP registration. Not sure why there isn't some fallback configuration or paging option.
I know a Cisco ATA would be the choice for this, but this is what I have at the moment :)
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If you think progress is ONLY one-way.
"Russians urged to return to landline phones
Amid widespread shutdowns of mobile internet services, the head of Russia’s largest telecommunications company has advised people to return to fixed-line #communication."#Security #Reliability #Resilience #Telephony #News #Technology #USA #Russia #Europe
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I'm looking for help with debugging a yate VOIP/SIP/RTP issue. I have a number setup to redirect to a sound file after a timeout (stand in for an voicemail IVR). When called from an internal number registered with yate it works. When called from the outside number (yate acting as a SIP client) it doesn't. Yate says it routed the call to the announcement but I don't see that reflected in the SIP/RTP traffic, the call keeps ringing until I hang up.
:BoostOK:
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I'm looking for help with debugging a yate VOIP/SIP/RTP issue. I have a number setup to redirect to a sound file after a timeout (stand in for an voicemail IVR). When called from an internal number registered with yate it works. When called from the outside number (yate acting as a SIP client) it doesn't. Yate says it routed the call to the announcement but I don't see that reflected in the SIP/RTP traffic, the call keeps ringing until I hang up.
:BoostOK:
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I'm looking for help with debugging a yate VOIP/SIP/RTP issue. I have a number setup to redirect to a sound file after a timeout (stand in for an voicemail IVR). When called from an internal number registered with yate it works. When called from the outside number (yate acting as a SIP client) it doesn't. Yate says it routed the call to the announcement but I don't see that reflected in the SIP/RTP traffic, the call keeps ringing until I hang up.
:BoostOK:
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While I am ready to DTMF, I fear that the rubber ring around the speaker fits onto a now-long-gone hand-held telephone receiver design.
#DTMF #retro #telephony -
Hardware I returned to Xfinity this afternoon. The Arris telephony modem was about 20 years old. The cable box / DVT was not much younger.
We dropped cable TV, VOIP, and stopped renting modems (bought my own Arris S33v3) very recently, saving hundreds of dollars per month on the bill.
#Xfinity #cable #cabletv #vintage #networking #vintagehardware #docsis #telephony #modem #modems
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Hardware I returned to Xfinity this afternoon. The Arris telephony modem was about 20 years old. The cable box / DVT was not much younger.
We dropped cable TV, VOIP, and stopped renting modems (bought my own Arris S33v3) very recently, saving hundreds of dollars per month on the bill.
#Xfinity #cable #cabletv #vintage #networking #vintagehardware #docsis #telephony #modem #modems
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Hardware I returned to Xfinity this afternoon. The Arris telephony modem was about 20 years old. The cable box / DVT was not much younger.
We dropped cable TV, VOIP, and stopped renting modems (bought my own Arris S33v3) very recently, saving hundreds of dollars per month on the bill.
#Xfinity #cable #cabletv #vintage #networking #vintagehardware #docsis #telephony #modem #modems
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Hardware I returned to Xfinity this afternoon. The Arris telephony modem was about 20 years old. The cable box / DVT was not much younger.
We dropped cable TV, VOIP, and stopped renting modems (bought my own Arris S33v3) very recently, saving hundreds of dollars per month on the bill.
#Xfinity #cable #cabletv #vintage #networking #vintagehardware #docsis #telephony #modem #modems
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Hardware I returned to Xfinity this afternoon. The Arris telephony modem was about 20 years old. The cable box / DVT was not much younger.
We dropped cable TV, VOIP, and stopped renting modems (bought my own Arris S33v3) very recently, saving hundreds of dollars per month on the bill.
#Xfinity #cable #cabletv #vintage #networking #vintagehardware #docsis #telephony #modem #modems
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Now the other two are ported - all the #analogue #trunks are ceased and all traffic to and from the #PSTN from this #PBX is now #VOIP (using #PJSIP trunks on #FreePBX )
I disconnected all the analogue lines as #Openreach leave battery on them, that way they show as RED alarm on #Asterisk and won't be selected for any calls (worst case is they return a busy trunk status and the route will go to the next trunk, but I've taken away all the analogue circuits from every PSTN route)
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Now the other two are ported - all the #analogue #trunks are ceased and all traffic to and from the #PSTN from this #PBX is now #VOIP (using #PJSIP trunks on #FreePBX )
I disconnected all the analogue lines as #Openreach leave battery on them, that way they show as RED alarm on #Asterisk and won't be selected for any calls (worst case is they return a busy trunk status and the route will go to the next trunk, but I've taken away all the analogue circuits from every PSTN route)
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Now the other two are ported - all the #analogue #trunks are ceased and all traffic to and from the #PSTN from this #PBX is now #VOIP (using #PJSIP trunks on #FreePBX )
I disconnected all the analogue lines as #Openreach leave battery on them, that way they show as RED alarm on #Asterisk and won't be selected for any calls (worst case is they return a busy trunk status and the route will go to the next trunk, but I've taken away all the analogue circuits from every PSTN route)
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Now the other two are ported - all the #analogue #trunks are ceased and all traffic to and from the #PSTN from this #PBX is now #VOIP (using #PJSIP trunks on #FreePBX )
I disconnected all the analogue lines as #Openreach leave battery on them, that way they show as RED alarm on #Asterisk and won't be selected for any calls (worst case is they return a busy trunk status and the route will go to the next trunk, but I've taken away all the analogue circuits from every PSTN route)
