astlinux-users — AstLinux Users Mailing List

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Another update:
I actually think I have fixed the problem. I removed the adaptive jitterbuffer from voicemail in the dialplan and I was able to make it happen again after over 30 attempts. I put it back and it cut off the first call.
Will implement out in the wild and see if it fixes the problem. Will let you know how I go.
PS sorry for all the emails.

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 5:51 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Scratch the last email. Had it that only one of them dropped out and the other kept working.

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 5:04 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Another update:
In my testing I had two phone calls going simultaneously (one from my mobile and one from my deskphone) and they both dropped out at virtually the same time.
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: app.c:2010 __ast_play_and_record: Error writing frame
    -- Recording was 0 seconds long but needs to be at least 1 - abandoning
….
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: app.c:2010 __ast_play_and_record: Error writing frame

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 3:13 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Update:
Using tcpdump I managed to do a packet capture as the problem is unfortunately occurring frequently enough to make this possible.
After looking at the pcap with Wireshark, the RTP stream looked fine and I could not find any empty RTP frames e.g. all had payload entries and were the same size. They were all G.711 PCMA encoded as well.

Surely it cant be a disk write issue otherwise I would probably be seeing other issues and its usually intermittent?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 2:24 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hmm the problem is still there after these changes and I have now stopped my upgrades until its fixed.
After posting on the forum, jcolp has responded with:

--------
The two cases for format_wav to return an error for writing is:

  1.  It was given a frame with no data in it
  2.  An error occurred when writing it to the disk

The first case would require probably orchestrating things and going through the complete media flow to determine where/how a frame with no data appeared.
--------

Any ideas where I would start my troubleshooting?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Wednesday, 14 June 2023 at 6:16 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Scratch the last email. Had it that only one of them dropped out and the other kept working.

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 5:04 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Another update:
In my testing I had two phone calls going simultaneously (one from my mobile and one from my deskphone) and they both dropped out at virtually the same time.
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: app.c:2010 __ast_play_and_record: Error writing frame
    -- Recording was 0 seconds long but needs to be at least 1 - abandoning
….
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: app.c:2010 __ast_play_and_record: Error writing frame

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 3:13 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Update:
Using tcpdump I managed to do a packet capture as the problem is unfortunately occurring frequently enough to make this possible.
After looking at the pcap with Wireshark, the RTP stream looked fine and I could not find any empty RTP frames e.g. all had payload entries and were the same size. They were all G.711 PCMA encoded as well.

Surely it cant be a disk write issue otherwise I would probably be seeing other issues and its usually intermittent?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 2:24 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hmm the problem is still there after these changes and I have now stopped my upgrades until its fixed.
After posting on the forum, jcolp has responded with:

--------
The two cases for format_wav to return an error for writing is:

  1.  It was given a frame with no data in it
  2.  An error occurred when writing it to the disk

The first case would require probably orchestrating things and going through the complete media flow to determine where/how a frame with no data appeared.
--------

Any ideas where I would start my troubleshooting?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Wednesday, 14 June 2023 at 6:16 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Another update:
In my testing I had two phone calls going simultaneously (one from my mobile and one from my deskphone) and they both dropped out at virtually the same time.
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:13] WARNING[8830][C-000000c4]: app.c:2010 __ast_play_and_record: Error writing frame
    -- Recording was 0 seconds long but needs to be at least 1 - abandoning
….
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: file.c:293 ast_writestream: Translated frame write failed
[Jun 22 16:58:30] WARNING[8806][C-000000c3]: app.c:2010 __ast_play_and_record: Error writing frame

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 3:13 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Update:
Using tcpdump I managed to do a packet capture as the problem is unfortunately occurring frequently enough to make this possible.
After looking at the pcap with Wireshark, the RTP stream looked fine and I could not find any empty RTP frames e.g. all had payload entries and were the same size. They were all G.711 PCMA encoded as well.

Surely it cant be a disk write issue otherwise I would probably be seeing other issues and its usually intermittent?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 2:24 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hmm the problem is still there after these changes and I have now stopped my upgrades until its fixed.
After posting on the forum, jcolp has responded with:

--------
The two cases for format_wav to return an error for writing is:

  1.  It was given a frame with no data in it
  2.  An error occurred when writing it to the disk

The first case would require probably orchestrating things and going through the complete media flow to determine where/how a frame with no data appeared.
--------

Any ideas where I would start my troubleshooting?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Wednesday, 14 June 2023 at 6:16 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Update:
Using tcpdump I managed to do a packet capture as the problem is unfortunately occurring frequently enough to make this possible.
After looking at the pcap with Wireshark, the RTP stream looked fine and I could not find any empty RTP frames e.g. all had payload entries and were the same size. They were all G.711 PCMA encoded as well.

Surely it cant be a disk write issue otherwise I would probably be seeing other issues and its usually intermittent?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Thursday, 22 June 2023 at 2:24 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hmm the problem is still there after these changes and I have now stopped my upgrades until its fixed.
After posting on the forum, jcolp has responded with:

--------
The two cases for format_wav to return an error for writing is:

  1.  It was given a frame with no data in it
  2.  An error occurred when writing it to the disk

The first case would require probably orchestrating things and going through the complete media flow to determine where/how a frame with no data appeared.
--------

Any ideas where I would start my troubleshooting?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Wednesday, 14 June 2023 at 6:16 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Hmm the problem is still there after these changes and I have now stopped my upgrades until its fixed.
After posting on the forum, jcolp has responded with:

--------
The two cases for format_wav to return an error for writing is:

  1.  It was given a frame with no data in it
  2.  An error occurred when writing it to the disk

The first case would require probably orchestrating things and going through the complete media flow to determine where/how a frame with no data appeared.
--------

Any ideas where I would start my troubleshooting?

Regards
Michael Knill


From: Michael Knill <mic...@ip...>
Date: Wednesday, 14 June 2023 at 6:16 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Running ipsec behind Astlinux

[Michael K. <mic...@ip...>]

Thanks Lonnie very much for your response.

Yes I had some suspicions that this was the issue however I tried to drop off one and reconnect the other unsuccessfully.
Unfortunately its not us configuring the client so not sure if they are using NATT ☹ but I think with the information provided we will be able to get this sorted.
Yes I never use IPsec.

Thanks again.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Thursday, 22 June 2023 at 12:10 am
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Running ipsec behind Astlinux
Hi Michael,

First, answering your followup question:

> (Actually if this works...) Do I need any firewall rules for this? I did have AH, ESP and UDP500/4500 NAT’d previously.

No you don't, the AIF ipsec-vpn plugin automatically opens ports for an AstLinux IPsec VPN endpoint as well as supporting forwarding NAT'ed IPsec traffic.  Since you don't have the AstLinux IPsec VPN enabled, the described "hack" is to to enable the plugin to support forwarding NAT'ed IPsec traffic.


> Interestingly I had a Cisco router working behind it fine but we couldn’t get the second VPN up.

Ahhh, that explains a lot.

Note that NAT works with UDP and TCP by using the inbound/outbound 'port' and inbound/outbound IP address to create a connection tracking hash table.  Clients behind NAT can use multiple UDP/TCP connections to the same public server since they will each use different ports via NAT at the edge.

Now with IPsec using ESP, a raw IP protocol, there are no ports for the NAT connection tracking to use for uniqueness.  As a result, only one IPsec ESP client connection can be established to the same public server behind NAT.  A second IPsec ESP client connection will fail as long as the NAT table has an active, previous IPsec ESP client connection.

The solution to this is to configure the IPsec server and client to use IPsec NATT (NAT Transversal) where the IPsec payload uses 4500/UDP instead of ESP.  In both cases IPsec IKE uses 500/UDP to negotiate the connection.

In summary (as I see it):

1) If your goal is to establish more than one IPsec ESP client connection to the *same* public server, the AIF ipsec-vpn plugin "hack" will not help you.

2) If you can use IPsec NATT (NAT Transversal), the AIF ipsec-vpn plugin "hack" is not needed, that should work with most any NAT router.

Lonnie

Or, just use WireGuard :-)




> On Jun 21, 2023, at 1:01 AM, Michael Knill <mic...@ip...> wrote:
>
> Thanks Lonnie. I will give it a try.
> Interestingly I had a Cisco router working behind it fine but we couldn’t get the second VPN up. We changed it out for a TP-Link router so the customer could manage themselves and that didn’t work at all.
>
> Regards
> Michael Knill
>
>
> From: Lonnie Abelbeck <li...@lo...>
> Date: Tuesday, 20 June 2023 at 11:44 pm
> To: AstLinux Users Mailing List <ast...@li...>
> Subject: Re: [Astlinux-users] Running ipsec behind Astlinux
>
> Hi Michael,
>
> Good question...
>
> It sounds like AstLinux needs to perform IPsec pass-through while the AstLinux IPsec VPN is not enabled.
>
> As a quick "hack", using the Network tab ...
>
> Firewall Plugins: [ ipsec-vpn ] - { Configure Plugin }
>
> Ignore the "*** Do Not Edit Below Here ***" note and set ENABLED=1 in the lower section, per this diff:
>
> -- diff --
>  # AstLinux specific mappings, either edit your /mnt/kd/rc.conf file
>  # or, use Network tab -> [IPsec Configuration] from the web interface.
>  # ------------------------------------------------------------------------------
>  # Indent script section so script variables won't be merged
>
> -  ENABLED=0
> +  ENABLED=1
>    IPSEC_ALLOWED_HOSTS="0/0"
>    IPSEC_VPN_NETS=""
>    IPSEC_NAT_TRAVERSAL=0
>    vpntype_ipsec=0
> -- diff --
>
> "Save Changes" and "Restart Firewall" to apply the change.
>
> Please report back if this solves your issue.
>
> BTW, alternatively, if the internal IPsec client was configured to use NAT Traversal, that should also work without AstLinux firewall tweaks.
>
> Lonnie
>
>
>
> > On Jun 20, 2023, at 3:19 AM, Michael Knill <mic...@ip...> wrote:
> >
> > Hi Group
> >
> > I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
> > What could be the problem? Are there any additional rules I need to add?
> >
> > This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.
> >
> > Regards
> >
> > Michael Knill
> > Managing Director
> >
> > D: +61 2 6189 1360
> > P: +61 2 6140 4656
> > E: mic...@ip...
> > W: ipcsolutions.com.au
> >
> >  <image001.png>
> > Smarter Business Communications
> >
> > _______________________________________________
> > Astlinux-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Running ipsec behind Astlinux

[Lonnie A. <li...@lo...>]

Hi Michael,

First, answering your followup question:

> (Actually if this works...) Do I need any firewall rules for this? I did have AH, ESP and UDP500/4500 NAT’d previously.

No you don't, the AIF ipsec-vpn plugin automatically opens ports for an AstLinux IPsec VPN endpoint as well as supporting forwarding NAT'ed IPsec traffic.  Since you don't have the AstLinux IPsec VPN enabled, the described "hack" is to to enable the plugin to support forwarding NAT'ed IPsec traffic.


> Interestingly I had a Cisco router working behind it fine but we couldn’t get the second VPN up.

Ahhh, that explains a lot.

Note that NAT works with UDP and TCP by using the inbound/outbound 'port' and inbound/outbound IP address to create a connection tracking hash table.  Clients behind NAT can use multiple UDP/TCP connections to the same public server since they will each use different ports via NAT at the edge.

Now with IPsec using ESP, a raw IP protocol, there are no ports for the NAT connection tracking to use for uniqueness.  As a result, only one IPsec ESP client connection can be established to the same public server behind NAT.  A second IPsec ESP client connection will fail as long as the NAT table has an active, previous IPsec ESP client connection.

The solution to this is to configure the IPsec server and client to use IPsec NATT (NAT Transversal) where the IPsec payload uses 4500/UDP instead of ESP.  In both cases IPsec IKE uses 500/UDP to negotiate the connection.

In summary (as I see it):

1) If your goal is to establish more than one IPsec ESP client connection to the *same* public server, the AIF ipsec-vpn plugin "hack" will not help you.

2) If you can use IPsec NATT (NAT Transversal), the AIF ipsec-vpn plugin "hack" is not needed, that should work with most any NAT router.

Lonnie

Or, just use WireGuard :-)




> On Jun 21, 2023, at 1:01 AM, Michael Knill <mic...@ip...> wrote:
> 
> Thanks Lonnie. I will give it a try.
> Interestingly I had a Cisco router working behind it fine but we couldn’t get the second VPN up. We changed it out for a TP-Link router so the customer could manage themselves and that didn’t work at all.
>  
> Regards
> Michael Knill
>  
>  
> From: Lonnie Abelbeck <li...@lo...>
> Date: Tuesday, 20 June 2023 at 11:44 pm
> To: AstLinux Users Mailing List <ast...@li...>
> Subject: Re: [Astlinux-users] Running ipsec behind Astlinux
> 
> Hi Michael,
> 
> Good question...
> 
> It sounds like AstLinux needs to perform IPsec pass-through while the AstLinux IPsec VPN is not enabled.
> 
> As a quick "hack", using the Network tab ...
> 
> Firewall Plugins: [ ipsec-vpn ] - { Configure Plugin }
> 
> Ignore the "*** Do Not Edit Below Here ***" note and set ENABLED=1 in the lower section, per this diff:
> 
> -- diff --
>  # AstLinux specific mappings, either edit your /mnt/kd/rc.conf file
>  # or, use Network tab -> [IPsec Configuration] from the web interface.
>  # ------------------------------------------------------------------------------
>  # Indent script section so script variables won't be merged
>  
> -  ENABLED=0
> +  ENABLED=1
>    IPSEC_ALLOWED_HOSTS="0/0"
>    IPSEC_VPN_NETS=""
>    IPSEC_NAT_TRAVERSAL=0
>    vpntype_ipsec=0
> -- diff --
> 
> "Save Changes" and "Restart Firewall" to apply the change.
> 
> Please report back if this solves your issue.
> 
> BTW, alternatively, if the internal IPsec client was configured to use NAT Traversal, that should also work without AstLinux firewall tweaks.
> 
> Lonnie
> 
> 
> 
> > On Jun 20, 2023, at 3:19 AM, Michael Knill <mic...@ip...> wrote:
> > 
> > Hi Group
> >  
> > I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
> > What could be the problem? Are there any additional rules I need to add?
> >  
> > This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.
> >  
> > Regards
> >  
> > Michael Knill
> > Managing Director
> >  
> > D: +61 2 6189 1360
> > P: +61 2 6140 4656
> > E: mic...@ip...
> > W: ipcsolutions.com.au
> >  
> >  <image001.png>
> > Smarter Business Communications
> >  
> > _______________________________________________
> > Astlinux-users mailing list
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > 
> > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
> 
> 
> 
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Running ipsec behind Astlinux

[Michael K. <mic...@ip...>]

Actually if this works, is there any reason why I could not have this implemented for all my systems?
Do I need any firewall rules for this? I did have AH, ESP and UDP500/4500 NAT’d previously.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 20 June 2023 at 11:44 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Running ipsec behind Astlinux
Hi Michael,

Good question...

It sounds like AstLinux needs to perform IPsec pass-through while the AstLinux IPsec VPN is not enabled.

As a quick "hack", using the Network tab ...

Firewall Plugins: [ ipsec-vpn ] - { Configure Plugin }

Ignore the "*** Do Not Edit Below Here ***" note and set ENABLED=1 in the lower section, per this diff:

-- diff --
 # AstLinux specific mappings, either edit your /mnt/kd/rc.conf file
 # or, use Network tab -> [IPsec Configuration] from the web interface.
 # ------------------------------------------------------------------------------
 # Indent script section so script variables won't be merged

-  ENABLED=0
+  ENABLED=1
   IPSEC_ALLOWED_HOSTS="0/0"
   IPSEC_VPN_NETS=""
   IPSEC_NAT_TRAVERSAL=0
   vpntype_ipsec=0
-- diff --

"Save Changes" and "Restart Firewall" to apply the change.

Please report back if this solves your issue.

BTW, alternatively, if the internal IPsec client was configured to use NAT Traversal, that should also work without AstLinux firewall tweaks.

Lonnie



> On Jun 20, 2023, at 3:19 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
> What could be the problem? Are there any additional rules I need to add?
>
> This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Running ipsec behind Astlinux

[Michael K. <mic...@ip...>]

Thanks Lonnie. I will give it a try.
Interestingly I had a Cisco router working behind it fine but we couldn’t get the second VPN up. We changed it out for a TP-Link router so the customer could manage themselves and that didn’t work at all.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 20 June 2023 at 11:44 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Running ipsec behind Astlinux
Hi Michael,

Good question...

It sounds like AstLinux needs to perform IPsec pass-through while the AstLinux IPsec VPN is not enabled.

As a quick "hack", using the Network tab ...

Firewall Plugins: [ ipsec-vpn ] - { Configure Plugin }

Ignore the "*** Do Not Edit Below Here ***" note and set ENABLED=1 in the lower section, per this diff:

-- diff --
 # AstLinux specific mappings, either edit your /mnt/kd/rc.conf file
 # or, use Network tab -> [IPsec Configuration] from the web interface.
 # ------------------------------------------------------------------------------
 # Indent script section so script variables won't be merged

-  ENABLED=0
+  ENABLED=1
   IPSEC_ALLOWED_HOSTS="0/0"
   IPSEC_VPN_NETS=""
   IPSEC_NAT_TRAVERSAL=0
   vpntype_ipsec=0
-- diff --

"Save Changes" and "Restart Firewall" to apply the change.

Please report back if this solves your issue.

BTW, alternatively, if the internal IPsec client was configured to use NAT Traversal, that should also work without AstLinux firewall tweaks.

Lonnie



> On Jun 20, 2023, at 3:19 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
> What could be the problem? Are there any additional rules I need to add?
>
> This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Running ipsec behind Astlinux

[Lonnie A. <li...@lo...>]

Hi Michael,

Good question...

It sounds like AstLinux needs to perform IPsec pass-through while the AstLinux IPsec VPN is not enabled.

As a quick "hack", using the Network tab ...

Firewall Plugins: [ ipsec-vpn ] - { Configure Plugin }

Ignore the "*** Do Not Edit Below Here ***" note and set ENABLED=1 in the lower section, per this diff:

-- diff --
 # AstLinux specific mappings, either edit your /mnt/kd/rc.conf file
 # or, use Network tab -> [IPsec Configuration] from the web interface.
 # ------------------------------------------------------------------------------
 # Indent script section so script variables won't be merged
 
-  ENABLED=0
+  ENABLED=1
   IPSEC_ALLOWED_HOSTS="0/0"
   IPSEC_VPN_NETS=""
   IPSEC_NAT_TRAVERSAL=0
   vpntype_ipsec=0
-- diff --

"Save Changes" and "Restart Firewall" to apply the change.

Please report back if this solves your issue.

BTW, alternatively, if the internal IPsec client was configured to use NAT Traversal, that should also work without AstLinux firewall tweaks.

Lonnie



> On Jun 20, 2023, at 3:19 AM, Michael Knill <mic...@ip...> wrote:
> 
> Hi Group
>  
> I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
> What could be the problem? Are there any additional rules I need to add?
>  
> This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.
>  
> Regards
>  
> Michael Knill
> Managing Director
>  
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>  
>  <image001.png>
> Smarter Business Communications
>  
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

[Astlinux-users] Running ipsec behind Astlinux

[Michael K. <mic...@ip...>]

Hi Group

I have an ipsec VPN device behind Astlinux and it cannot connect. When I stick the device behind a 4G enabled Mikrotik router then it works fine.
What could be the problem? Are there any additional rules I need to add?

This is certainly very annoying and hopefully I can fix it before it uses up all my 4G data.

Regards

Michael Knill
Managing Director

D: +61 2 6189 1360<tel:+61261891360>
P: +61 2 6140 4656<tel:+61261404656>
E: mic...@ip...<mailto:mic...@ip...>
W: ipcsolutions.com.au<https://ipcsolutions.com.au/>

 [Icon  Description automatically generated]
Smarter Business Communications

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Hi Lonnie

Thanks for this.

So the testing I performed was to call into the system from my mobile to a number that goes directly to voicemail without a greeting. If the call stayed up for a couple of seconds then I would hang up and call again.
Whenever I got the Warning messages, the call actually dropped. Another log line I didnt add was:
-- Recording was 0 seconds long but needs to be at least 1 – abandoning

I also posted on the Asterisk forum and someone mentioned that a solution to the problem could be setting “transmit_silence=yes” in asterisk.conf which I tried and it significantly reduced (possibly eliminated) the problem.

I will try setting this at a couple of our problem sites to see if it fixes the problem and let you know how I go.

Regards
Michael Knill


From: Lonnie Abelbeck <li...@lo...>
Date: Tuesday, 13 June 2023 at 10:26 pm
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7
Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>
> Any ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
>  <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....



_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Lonnie A. <li...@lo...>]

Hi Michael,

I looked through the Asterisk code, this is basic core code, but some 'code stirring' has occurred between 13 and 16.

If you can replicate it in the lab, does Astlinux 1.5.0 / 13se work as expected with your voicemail.conf?

Does the error occur only on long (longer) voicemails?

Does the error occur intermittently or all the time?  Any pattern?

Lonnie



> On Jun 13, 2023, at 5:52 AM, Michael Knill <mic...@ip...> wrote:
> 
> Hi Group
>  
> Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
> Im intermittently getting the following which from testing happens prior to it dropping out:
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
> Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame
>  
> Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
> [general]
> format = wav
> maxsecs = 180
> minsecs = 1
> maxmsg = 1000
> maxgreet = 60
> maxsilence = 0
> minpassword = 4
> silencethreshold = 128
> maxlogins = 3
> nextaftercmd = yes
> sendvoicemail = yes
> review = yes
> operator = yes
> forcename = yes
> forcegreetings = yes
> tempgreetwarn = yes
> callback = DialPlan1
> exitcontext = voicemail-exit
> externpass = /mnt/kd/scripts/vm_password_sync
> externnotify = php /mnt/kd/scripts/voicemailnotify.php
>  
> I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.
>  
> Any ideas?
>  
> Regards
>  
> Michael Knill
> Managing Director
>  
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>  
>  <image001.png>
> Smarter Business Communications
>  
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

[Astlinux-users] Problems with voicemail and Asterisk 16 on Astlinux 1.4.7

[Michael K. <mic...@ip...>]

Hi Group

Im trying to find out why I am getting voicemail errors on Asterisk 16 on Astlinux 1.4.7 and hoping someone may have an idea where I should start investigating. Im getting reports and example voicemails where the person has been cut off mid recording only on Asterisk 16 on Astlinux 1.4.7.
Im intermittently getting the following which from testing happens prior to it dropping out:
Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: file.c:293 in ast_writestream: Translated frame write failed
Jun 13 13:18:47 25160-Clinic88-CM1 local0.warn asterisk[1203]: WARNING[1533][C-000004bc]: app.c:2010 in __ast_play_and_record: Error writing frame

Nothing on 1.3.10 using Asterisk 13. Both have the same voicemail config:
[general]
format = wav
maxsecs = 180
minsecs = 1
maxmsg = 1000
maxgreet = 60
maxsilence = 0
minpassword = 4
silencethreshold = 128
maxlogins = 3
nextaftercmd = yes
sendvoicemail = yes
review = yes
operator = yes
forcename = yes
forcegreetings = yes
tempgreetwarn = yes
callback = DialPlan1
exitcontext = voicemail-exit
externpass = /mnt/kd/scripts/vm_password_sync
externnotify = php /mnt/kd/scripts/voicemailnotify.php

I have tried Astlinux 1.5.0 and it still happens. I cant seem to find any related bugs.

Any ideas?

Regards

Michael Knill
Managing Director

D: +61 2 6189 1360<tel:+61261891360>
P: +61 2 6140 4656<tel:+61261404656>
E: mic...@ip...<mailto:mic...@ip...>
W: ipcsolutions.com.au<https://ipcsolutions.com.au/>

 [Icon  Description automatically generated]
Smarter Business Communications

[Astlinux-users] AstLinux Pre-Release: astlinux-1.5-5809-61d23a

[Lonnie A. <li...@lo...>]

Announcing AstLinux Pre-Release: astlinux-1.5-5809-61d23a

AstLinux Project -> Development
https://www.astlinux-project.org/dev.html

** The AstLinux Team is regularly upgrading packages containing security and bug fixes as well as adding new features of our own.

-- Linux Kernel 5.10.178 (version bump), security and bug fixes

-- OpenSSL, version bump to 1.1.1u, security fixes: CVE-2023-0464, CVE-2023-0465, CVE-2023-0466, CVE-2023-2650

-- libcurl (curl) version bump to 8.1.2, security fixes: CVE-2023-27535, CVE-2023-28319, etc.

-- dnsmasq, version 2.84, security fix: CVE-2023-28450

-- Fossil, (major) version bump to 2.22

-- keepalived, version bump to 2.2.8

-- libpcap, version bump to 1.10.4

-- libxml2, version bump to 2.10.4, security fixes: CVE-2023-29469, CVE-2023-28484

-- ncurses, version bump to 6.4, security fix: CVE-2023-29491 using --disable-root-environ build

-- pciutils, version bump to 3.10.0

-- sqlite, version bump to 3.42.0

-- sngrep, version bump to 1.7.0

-- tcpdump, version 4.99.4

-- udev (eudev), version bump to 3.2.12

-- zabbix, version bump to 4.0.46

-- ca-certificates, update trusted root certificates 2023-05-30

-- mac2vendor, oui.txt database snapshot 2023-06-10

-- Time Zone Database update, tzdata2023c and php-timezonedb-2023.3

-- Asterisk 13.38.3 ('13se' no change)
	Last Asterisk 13.x "Legacy" version, built --without-pjproject

-- Asterisk 16.30.0 (no change) and 18.18.0 (version bump)
	Add Asterisk 16.x pjsip_pubsub-fixes-for-pjsip-2.13 patch

-- DAHDI, dahdi-linux 3.2.0 (no change) and dahdi-tools 3.2.0 (no change)

-- pjsip 2.13 (version bump)

-- Complete Pre-Release ChangeLog:
https://astlinux-project.org/beta/astlinux-changelog/ChangeLog.txt

The "AstLinux Pre-Release ChangeLog" and "Pre-Release Repository URL" entries can be found under the "Development" tab of the AstLinux Project web site ...

AstLinux Project -> Development
https://www.astlinux-project.org/dev.html


AstLinux Team

Re: [Astlinux-users] Astlinux reliability

[Lonnie A. <li...@lo...>]

> On Jun 9, 2023, at 9:54 PM, Michael Knill <mic...@ip...> wrote:
> 
> System Uptime: 989 days, 1:29
>  
> Its on an APU2 in a hospital environment so never had a power failure.
> Yes I should have upgraded it long ago but pretty cool!
>  
> Regards
> Michael Knill

Hey Michael,

Thanks for reporting!

Over the years, I also have witnessed multi-year uptimes for special situation AstLinux boxes.

Lonnie

[Astlinux-users] Astlinux reliability

[Michael K. <mic...@ip...>]

System Uptime: 989 days, 1:29

Its on an APU2 in a hospital environment so never had a power failure.
Yes I should have upgraded it long ago but pretty cool!

Regards
Michael Knill

Re: [Astlinux-users] EXTIF_ALIAS

[Lonnie A. <li...@lo...>]

Hi David,

To be clear the variable name is 'EXTIP_ALIAS'
--
## External Interface Alias (Virtual) IPv4 Addresses
## If EXTIP (or EXT2IP) is set, using a 'static' configuration, alias interfaces
## on EXTIF (or EXT2IF) may be defined creating $EXTIF:1, $EXTIF:2, etc. .
## Multiple IPv4 addresses are space separated.
#EXTIP_ALIAS="192.168.25.3 192.168.25.4"
#EXT2IP_ALIAS="192.168.25.3 192.168.25.4"
--

This was added to support certain business level ISPs that allowed more than one static IP address on the external interface.  This would be used via the firewall 'NAT EXT->LAN' and the 'NAT EXT' setting to selectively NAT inbound traffic to LAN devices for multiple static external IPv4 addresses.



Keep in mind this only applies to static external IP addresses provided by your ISP.  Also the static external link setting (and any /mnt/kd/rc.elocal added routes) should be maintained with with the link cycling ... unlike with DHCP where the IP/routes are cleared/changed on a link cycling.

> I want to add an alias of 192.168.100.xx/24 to my external interface, with that I can access 192.168.100.1 which is the IP address of my (everyone's?) cable modem.

Hmm, I'm not sure why you would need that.  I personally can reach my cable modem at https://192.168.100.1/ from my LAN.

If you changed the firewall defaults, such as set the firewall RESERVED_NET_DROP setting to "1" in user.conf, that would block 192.168.100.1 access.

Lonnie



> On Jun 8, 2023, at 9:28 AM, David Kerr <Da...@Ke...> wrote:
> 
> Astlinux network initialization script has the ability to add an additional IP address to external interfaces.  You can define a list of IP addresses in the EXTIF_ALIAS and EXT2IF_ALIAS variables in user.conf.  However the script is hard coded to apply a /32 network mask.  Was this deliberate?
> 
> I want to add an alias of 192.168.100.xx/24 to my external interface, with that I can access 192.168.100.1 which is the IP address of my (everyone's?) cable modem.  I have been doing this manually in rc.local but discovered that this is not resilient to the link going down/up, which is when I discovered that the network script has this alias support.  But the /32 netmask prevents routing to any other devices because the subnet is, well, zero in length.
> 
> It feels like the network script should either require that the netmask is included in the EXTIF_ALIAS, or test to see if one is specified and only add /32 if none is provided (I suggest /32 for backward compatibility only... I think it should have defaulted to /24).
> 
> Thoughts?
> David 
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

[Astlinux-users] EXTIF_ALIAS

[David K. <da...@ke...>]

Astlinux network initialization script has the ability to add an additional
IP address to external interfaces.  You can define a list of IP addresses
in the EXTIF_ALIAS and EXT2IF_ALIAS variables in user.conf.  However the
script is hard coded to apply a /32 network mask.  Was this deliberate?

I want to add an alias of 192.168.100.xx/24 to my external interface,
with that I can access 192.168.100.1 which is the IP address of my
(everyone's?) cable modem.  I have been doing this manually in rc.local but
discovered that this is not resilient to the link going down/up, which is
when I discovered that the network script has this alias support.  But the
/32 netmask prevents routing to any other devices because the subnet is,
well, zero in length.

It feels like the network script should either require that the netmask is
included in the EXTIF_ALIAS, or test to see if one is specified and only
add /32 if none is provided (I suggest /32 for backward compatibility
only... I think it should have defaulted to /24).

Thoughts?
David

Re: [Astlinux-users] Hello all and I need help.

[Lonnie A. <li...@lo...>]

Yup, and the PCI-ids removal here:

https://github.com/asterisk/dahdi-linux/commit/29cb229cd3f1d252872b7f1924b6e3be941f7ad3

Lonnie


> On Jun 5, 2023, at 8:51 AM, Michael Keuter <li...@mk...> wrote:
> 
> The references were removed in 2018:
> 
> https://github.com/asterisk/dahdi-linux/commit/75620dd9ef6ac746745a1ecab4ef925a5b9e2988
> 
>> Am 05.06.2023 um 15:11 schrieb Lonnie Abelbeck <li...@lo...>:
>> 
>> Hi Tahiro,
>> 
>> A quick look at the dahdi-linux code for the "wcb4xxp", the PCI-ids appear to be defined here [1]
>> 
>> But my gut feeling, there may be more to it than simply adding a line to DEFINE_PCI_DEVICE_TABLE
>> 
>> Question, are Digium wcb4xxp ISDN cards available on the used market at a reasonable price?
>> 
>> Lonnie
>> 
>> [1] https://github.com/asterisk/dahdi-linux/blob/4397c55319154a8dc89022f6f75c683d6af12d54/drivers/dahdi/wcb4xxp/base.c#LL3642C1-L3651C3
>> 
>> 
>> 
>>> On Jun 5, 2023, at 2:51 AM, Tahiro Hashizume via Astlinux-users <ast...@li...> wrote:
>>> 
>>> Alright, I hope the attached TXT file clarifies the situation I am facing.
>>> Looks to me like the driver in question does not have the necessary PCI-ID listed.
>>> 
>>> On Thu, Jun 1, 2023 at 4:33 PM Tahiro Hashizume <ta...@ha...> wrote:
>>> Hi Michael,
>>> 
>>> The telephone service provided by the local telco (NTT East of Japan) is based on a very standard format (SIP+RTSP with SIP session timer of 300 seconds) provided on a closed network over fiber.
>>> Yet I still consider the interfacing of asterisk with the local telco via ISDN to be a valid option for the following reasons:
>>> A.) The information required for SIP registration incl. account, domain and SIP server address(es) are provided via vendor-specific options of DHCPv4+DHCPv6-PD.
>>> Given the format of the service (as mentioned above), direct-interfacing of asterisk with telco is no rocket science in principle, but doing so with some reliability is another thing.
>>> While there is sip-proxy software (non-OSS) available for Linux which also functions as a DHCP client, I find it rather silly to use it.
>>> B.)The local telco also requires that any non-hardware/non-certified IP-PBXes directly interfaced with their VoIP servers via IPv4/IPv6 to be inspected for security and compatibility. In terms of direct-interfacing asterisk with the telco, this means having asterisk config files checked by telco's engineers (AND THE INSPECTION COSTS A LOT!!!). The aforementioned sip-proxy is certified-compatible with the telco and effectively eliminates the need for inspection.
>>> 
>>> Given that B400P available through the local distributor is a telco-certified device and the telco also provides a ISDN gateway for the service (which has either two or four BRIs and ethernet), ISDN-interfacing of asterisk is a seemingly decent choice. Yes, it's a problem so easy to solve in principle but not so in reality.
>>> 
>>> Now, the card is listed in lspci, but is not visible from DAHDI utilities. My guess is that it's due to the PCI VID&PID of B400P that is not listed in "modinfo wcb4xxp". Documentation by OpenVox also says that a little patching is necessary, so things make sense overall.
>>> 
>>> I'll include the PID and VID with the next email should there be a demand for it.
>>> 
>>> Any comments and ideas are appreciated.
>>> 
>>> Tahiro
>>> 
>>> On Mon, May 29, 2023 at 6:38 PM Michael Keuter <li...@mk...> wrote:
>>> 
>>> 
>>>> Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <ast...@li...>:
>>>> 
>>>> Dear whom it may concern.
>>>> 
>>>> I've recently got my hands on a OpenVox B400P ISDN BRI card.
>>>> It seems that DAHDI included with Astlinux isn't built to support the card and I'm now trying to figure out how to build the image with the support included.
>>>> It's been a while since I started fiddling with OSS and I have been fairly comfortable building stuff from sources although I am not yet able to write my own Makefile and so on.
>>>> Any ideas on how I should get started?
>>>> 
>>>> P.S.-I have managed to build the toolchain and Astlinux image by default config for Asterisk 18.x. 
>>>> 
>>>> Regards.
>>> 
>>> Hi Tahiro,
>>> 
>>> the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.
>>> 
>>> https://doc.astlinux-project.org/userdoc:dahdi
>>> 
>>> So in principle this should work for your card:
>>> 
>>> DAHDIMODS="wcb4xxp dahdi_echocan_oslec"
>>> 
>>> I have switched all my ISDN based installations to berofix cards/boxes over 10 years ago.
>>> And now none is still in production :-). 
>>> 
>>> Michael
>>> 
>>> http://www.mksolutions.info
> 
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] Hello all and I need help.

[Michael K. <li...@mk...>]

The references were removed in 2018:

https://github.com/asterisk/dahdi-linux/commit/75620dd9ef6ac746745a1ecab4ef925a5b9e2988

> Am 05.06.2023 um 15:11 schrieb Lonnie Abelbeck <li...@lo...>:
> 
> Hi Tahiro,
> 
> A quick look at the dahdi-linux code for the "wcb4xxp", the PCI-ids appear to be defined here [1]
> 
> But my gut feeling, there may be more to it than simply adding a line to DEFINE_PCI_DEVICE_TABLE
> 
> Question, are Digium wcb4xxp ISDN cards available on the used market at a reasonable price?
> 
> Lonnie
> 
> [1] https://github.com/asterisk/dahdi-linux/blob/4397c55319154a8dc89022f6f75c683d6af12d54/drivers/dahdi/wcb4xxp/base.c#LL3642C1-L3651C3
> 
> 
> 
>> On Jun 5, 2023, at 2:51 AM, Tahiro Hashizume via Astlinux-users <ast...@li...> wrote:
>> 
>> Alright, I hope the attached TXT file clarifies the situation I am facing.
>> Looks to me like the driver in question does not have the necessary PCI-ID listed.
>> 
>> On Thu, Jun 1, 2023 at 4:33 PM Tahiro Hashizume <ta...@ha...> wrote:
>> Hi Michael,
>> 
>> The telephone service provided by the local telco (NTT East of Japan) is based on a very standard format (SIP+RTSP with SIP session timer of 300 seconds) provided on a closed network over fiber.
>> Yet I still consider the interfacing of asterisk with the local telco via ISDN to be a valid option for the following reasons:
>> A.) The information required for SIP registration incl. account, domain and SIP server address(es) are provided via vendor-specific options of DHCPv4+DHCPv6-PD.
>> Given the format of the service (as mentioned above), direct-interfacing of asterisk with telco is no rocket science in principle, but doing so with some reliability is another thing.
>> While there is sip-proxy software (non-OSS) available for Linux which also functions as a DHCP client, I find it rather silly to use it.
>> B.)The local telco also requires that any non-hardware/non-certified IP-PBXes directly interfaced with their VoIP servers via IPv4/IPv6 to be inspected for security and compatibility. In terms of direct-interfacing asterisk with the telco, this means having asterisk config files checked by telco's engineers (AND THE INSPECTION COSTS A LOT!!!). The aforementioned sip-proxy is certified-compatible with the telco and effectively eliminates the need for inspection.
>> 
>> Given that B400P available through the local distributor is a telco-certified device and the telco also provides a ISDN gateway for the service (which has either two or four BRIs and ethernet), ISDN-interfacing of asterisk is a seemingly decent choice. Yes, it's a problem so easy to solve in principle but not so in reality.
>> 
>> Now, the card is listed in lspci, but is not visible from DAHDI utilities. My guess is that it's due to the PCI VID&PID of B400P that is not listed in "modinfo wcb4xxp". Documentation by OpenVox also says that a little patching is necessary, so things make sense overall.
>> 
>> I'll include the PID and VID with the next email should there be a demand for it.
>> 
>> Any comments and ideas are appreciated.
>> 
>> Tahiro
>> 
>> On Mon, May 29, 2023 at 6:38 PM Michael Keuter <li...@mk...> wrote:
>> 
>> 
>>> Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <ast...@li...>:
>>> 
>>> Dear whom it may concern.
>>> 
>>> I've recently got my hands on a OpenVox B400P ISDN BRI card.
>>> It seems that DAHDI included with Astlinux isn't built to support the card and I'm now trying to figure out how to build the image with the support included.
>>> It's been a while since I started fiddling with OSS and I have been fairly comfortable building stuff from sources although I am not yet able to write my own Makefile and so on.
>>> Any ideas on how I should get started?
>>> 
>>> P.S.-I have managed to build the toolchain and Astlinux image by default config for Asterisk 18.x. 
>>> 
>>> Regards.
>> 
>> Hi Tahiro,
>> 
>> the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.
>> 
>> https://doc.astlinux-project.org/userdoc:dahdi
>> 
>> So in principle this should work for your card:
>> 
>> DAHDIMODS="wcb4xxp dahdi_echocan_oslec"
>> 
>> I have switched all my ISDN based installations to berofix cards/boxes over 10 years ago.
>> And now none is still in production :-). 
>> 
>> Michael
>> 
>> http://www.mksolutions.info


Michael

http://www.mksolutions.info

Re: [Astlinux-users] Hello all and I need help.

[Lonnie A. <li...@lo...>]

Hi Tahiro,

A quick look at the dahdi-linux code for the "wcb4xxp", the PCI-ids appear to be defined here [1]

But my gut feeling, there may be more to it than simply adding a line to DEFINE_PCI_DEVICE_TABLE

Question, are Digium wcb4xxp ISDN cards available on the used market at a reasonable price?

Lonnie

[1] https://github.com/asterisk/dahdi-linux/blob/4397c55319154a8dc89022f6f75c683d6af12d54/drivers/dahdi/wcb4xxp/base.c#LL3642C1-L3651C3



> On Jun 5, 2023, at 2:51 AM, Tahiro Hashizume via Astlinux-users <ast...@li...> wrote:
> 
> Alright, I hope the attached TXT file clarifies the situation I am facing.
> Looks to me like the driver in question does not have the necessary PCI-ID listed.
> 
> On Thu, Jun 1, 2023 at 4:33 PM Tahiro Hashizume <ta...@ha...> wrote:
> Hi Michael,
> 
> The telephone service provided by the local telco (NTT East of Japan) is based on a very standard format (SIP+RTSP with SIP session timer of 300 seconds) provided on a closed network over fiber.
> Yet I still consider the interfacing of asterisk with the local telco via ISDN to be a valid option for the following reasons:
> A.) The information required for SIP registration incl. account, domain and SIP server address(es) are provided via vendor-specific options of DHCPv4+DHCPv6-PD.
> Given the format of the service (as mentioned above), direct-interfacing of asterisk with telco is no rocket science in principle, but doing so with some reliability is another thing.
> While there is sip-proxy software (non-OSS) available for Linux which also functions as a DHCP client, I find it rather silly to use it.
> B.)The local telco also requires that any non-hardware/non-certified IP-PBXes directly interfaced with their VoIP servers via IPv4/IPv6 to be inspected for security and compatibility. In terms of direct-interfacing asterisk with the telco, this means having asterisk config files checked by telco's engineers (AND THE INSPECTION COSTS A LOT!!!). The aforementioned sip-proxy is certified-compatible with the telco and effectively eliminates the need for inspection.
> 
> Given that B400P available through the local distributor is a telco-certified device and the telco also provides a ISDN gateway for the service (which has either two or four BRIs and ethernet), ISDN-interfacing of asterisk is a seemingly decent choice. Yes, it's a problem so easy to solve in principle but not so in reality.
> 
> Now, the card is listed in lspci, but is not visible from DAHDI utilities. My guess is that it's due to the PCI VID&PID of B400P that is not listed in "modinfo wcb4xxp". Documentation by OpenVox also says that a little patching is necessary, so things make sense overall.
> 
> I'll include the PID and VID with the next email should there be a demand for it.
> 
> Any comments and ideas are appreciated.
> 
> Tahiro
> 
> On Mon, May 29, 2023 at 6:38 PM Michael Keuter <li...@mk...> wrote:
> 
> 
> > Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <ast...@li...>:
> > 
> > Dear whom it may concern.
> > 
> > I've recently got my hands on a OpenVox B400P ISDN BRI card.
> > It seems that DAHDI included with Astlinux isn't built to support the card and I'm now trying to figure out how to build the image with the support included.
> > It's been a while since I started fiddling with OSS and I have been fairly comfortable building stuff from sources although I am not yet able to write my own Makefile and so on.
> > Any ideas on how I should get started?
> > 
> > P.S.-I have managed to build the toolchain and Astlinux image by default config for Asterisk 18.x. 
> > 
> > Regards.
> 
> Hi Tahiro,
> 
> the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.
> 
> https://doc.astlinux-project.org/userdoc:dahdi
> 
> So in principle this should work for your card:
> 
> DAHDIMODS="wcb4xxp dahdi_echocan_oslec"
> 
> I have switched all my ISDN based installations to berofix cards/boxes over 10 years ago.
> And now none is still in production :-). 
> 
> Michael
> 
> http://www.mksolutions.info

Re: [Astlinux-users] Hello all and I need help.

[Tahiro H. <ta...@ha...>]

Alright, I hope the attached TXT file clarifies the situation I am facing.
Looks to me like the driver in question does not have the necessary PCI-ID
listed.

On Thu, Jun 1, 2023 at 4:33 PM Tahiro Hashizume <ta...@ha...>
wrote:

> Hi Michael,
> The telephone service provided by the local telco (NTT East of Japan) is
> based on a very standard format (SIP+RTSP with SIP session timer of 300
> seconds) provided on a closed network over fiber.
> Yet I still consider the interfacing of asterisk with the local telco via
> ISDN to be a valid option for the following reasons:
> A.) The information required for SIP registration incl. account, domain
> and SIP server address(es) are provided via vendor-specific options of
> DHCPv4+DHCPv6-PD.
> Given the format of the service (as mentioned above), direct-interfacing
> of asterisk with telco is no rocket science in principle, but doing so with
> some reliability is another thing.
> While there is sip-proxy software (non-OSS) available for Linux which also
> functions as a DHCP client, I find it rather silly to use it.
> B.)The local telco also requires that any non-hardware/non-certified
> IP-PBXes directly interfaced with their VoIP servers via IPv4/IPv6 to be
> inspected for security and compatibility. In terms of direct-interfacing
> asterisk with the telco, this means having asterisk config files checked by
> telco's engineers (AND THE INSPECTION COSTS A LOT!!!). The aforementioned
> sip-proxy is certified-compatible with the telco and effectively eliminates
> the need for inspection.
>
> Given that B400P available through the local distributor is a
> telco-certified device and the telco also provides a ISDN gateway for the
> service (which has either two or four BRIs and ethernet), ISDN-interfacing
> of asterisk is a seemingly decent choice. Yes, it's a problem so easy to
> solve in principle but not so in reality.
>
> Now, the card is listed in lspci, but is not visible from DAHDI utilities.
> My guess is that it's due to the PCI VID&PID of B400P that is not listed in
> "modinfo wcb4xxp". Documentation by OpenVox also says that a little
> patching is necessary, so things make sense overall.
>
> I'll include the PID and VID with the next email should there be a demand
> for it.
>
> Any comments and ideas are appreciated.
>
> Tahiro
>
> On Mon, May 29, 2023 at 6:38 PM Michael Keuter <li...@mk...>
> wrote:
>
>>
>>
>> > Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <
>> ast...@li...>:
>> >
>> > Dear whom it may concern.
>> >
>> > I've recently got my hands on a OpenVox B400P ISDN BRI card.
>> > It seems that DAHDI included with Astlinux isn't built to support the
>> card and I'm now trying to figure out how to build the image with the
>> support included.
>> > It's been a while since I started fiddling with OSS and I have been
>> fairly comfortable building stuff from sources although I am not yet able
>> to write my own Makefile and so on.
>> > Any ideas on how I should get started?
>> >
>> > P.S.-I have managed to build the toolchain and Astlinux image by
>> default config for Asterisk 18.x.
>> >
>> > Regards.
>>
>> Hi Tahiro,
>>
>> the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.
>>
>> https://doc.astlinux-project.org/userdoc:dahdi
>>
>> So in principle this should work for your card:
>>
>> DAHDIMODS="wcb4xxp dahdi_echocan_oslec"
>>
>> I have switched all my ISDN based installations to berofix cards/boxes
>> over 10 years ago.
>> And now none is still in production :-).
>>
>> Michael
>>
>> http://www.mksolutions.info
>>
>>
>>
>>
>>
>> _______________________________________________
>> Astlinux-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to
>> pa...@kr....
>>
>

Re: [Astlinux-users] Hello all and I need help.

[Tahiro H. <ta...@ha...>]

Hi Michael,
The telephone service provided by the local telco (NTT East of Japan) is
based on a very standard format (SIP+RTSP with SIP session timer of 300
seconds) provided on a closed network over fiber.
Yet I still consider the interfacing of asterisk with the local telco via
ISDN to be a valid option for the following reasons:
A.) The information required for SIP registration incl. account, domain and
SIP server address(es) are provided via vendor-specific options of
DHCPv4+DHCPv6-PD.
Given the format of the service (as mentioned above), direct-interfacing of
asterisk with telco is no rocket science in principle, but doing so with
some reliability is another thing.
While there is sip-proxy software (non-OSS) available for Linux which also
functions as a DHCP client, I find it rather silly to use it.
B.)The local telco also requires that any non-hardware/non-certified
IP-PBXes directly interfaced with their VoIP servers via IPv4/IPv6 to be
inspected for security and compatibility. In terms of direct-interfacing
asterisk with the telco, this means having asterisk config files checked by
telco's engineers (AND THE INSPECTION COSTS A LOT!!!). The aforementioned
sip-proxy is certified-compatible with the telco and effectively eliminates
the need for inspection.

Given that B400P available through the local distributor is a
telco-certified device and the telco also provides a ISDN gateway for the
service (which has either two or four BRIs and ethernet), ISDN-interfacing
of asterisk is a seemingly decent choice. Yes, it's a problem so easy to
solve in principle but not so in reality.

Now, the card is listed in lspci, but is not visible from DAHDI utilities.
My guess is that it's due to the PCI VID&PID of B400P that is not listed in
"modinfo wcb4xxp". Documentation by OpenVox also says that a little
patching is necessary, so things make sense overall.

I'll include the PID and VID with the next email should there be a demand
for it.

Any comments and ideas are appreciated.

Tahiro

On Mon, May 29, 2023 at 6:38 PM Michael Keuter <li...@mk...>
wrote:

>
>
> > Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <
> ast...@li...>:
> >
> > Dear whom it may concern.
> >
> > I've recently got my hands on a OpenVox B400P ISDN BRI card.
> > It seems that DAHDI included with Astlinux isn't built to support the
> card and I'm now trying to figure out how to build the image with the
> support included.
> > It's been a while since I started fiddling with OSS and I have been
> fairly comfortable building stuff from sources although I am not yet able
> to write my own Makefile and so on.
> > Any ideas on how I should get started?
> >
> > P.S.-I have managed to build the toolchain and Astlinux image by default
> config for Asterisk 18.x.
> >
> > Regards.
>
> Hi Tahiro,
>
> the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.
>
> https://doc.astlinux-project.org/userdoc:dahdi
>
> So in principle this should work for your card:
>
> DAHDIMODS="wcb4xxp dahdi_echocan_oslec"
>
> I have switched all my ISDN based installations to berofix cards/boxes
> over 10 years ago.
> And now none is still in production :-).
>
> Michael
>
> http://www.mksolutions.info
>
>
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] Hello all and I need help.

[Michael K. <li...@mk...>]

> Am 29.05.2023 um 07:39 schrieb Tahiro Hashizume via Astlinux-users <ast...@li...>:
> 
> Dear whom it may concern.
> 
> I've recently got my hands on a OpenVox B400P ISDN BRI card.
> It seems that DAHDI included with Astlinux isn't built to support the card and I'm now trying to figure out how to build the image with the support included.
> It's been a while since I started fiddling with OSS and I have been fairly comfortable building stuff from sources although I am not yet able to write my own Makefile and so on.
> Any ideas on how I should get started?
> 
> P.S.-I have managed to build the toolchain and Astlinux image by default config for Asterisk 18.x. 
> 
> Regards.

Hi Tahiro,

the only BRI driver in DAHDI is the WCB4XXP for 2-8 port HFS-chip cards.

https://doc.astlinux-project.org/userdoc:dahdi

So in principle this should work for your card:

DAHDIMODS="wcb4xxp dahdi_echocan_oslec"

I have switched all my ISDN based installations to berofix cards/boxes over 10 years ago.
And now none is still in production :-). 

Michael

http://www.mksolutions.info