Autopatch

So since I am on a roll today, here’s another to Chew on.

I have Incredible PBX setup on a Pi here. I use it (at least till Google shut off the GVSIP service today) for my home phone as well as Allstar and Hamshack Hotline (See hamshackhotline.com for more info).

I want to make one of my allstar nodes dial out through my PBX…Incredible PBX is based on FreePBX and Asterisk sooo…

I have calling from the PBX to ASL working fine. I can dial an extension from any of my phones and it connects to my ASL nodes.

I am stumped on From ASL to PBX. Here is what I have done…

I followed the Directions on this page: http://docs.allstarlink.org/drupal/node/133

So first I went into Rpt.conf and set those settings. I used "pstn-out"for the context in Rpt.conf for the autopatch. Next I went into extensions.conf and set the following in pstn-out:

[pstn-out]

exten => _XXX,1,Dial,SIP/pbx-autopatch/${Exten}

exten => _XXX,2, congestion

Note that XXX is for internal extensions. I use 3 digit extensions internally for phones. Next I went to sip.conf and set up the following stanza:

[pbx-autopatch]

type=peer

allowguest=yes

nat=yes

autocreatepeer=yes

insecure=port,invite

auth=md5

username=

password=nunya

host=

qualify=no

disallow=all

allow=ulaw

allowexternaldomains=yes

context=pstn-out

When I dial *61 + internal extension (203 for example) it waits a few seconds then gives me fast busy

When I watch on the asterisk CLI it tries to connect is not able to connect and at one point says “no such host: pbx-autopatch”

(I would paste the log but I am not home to be able to copy and paste)

Any initial thoughts? I’ve been playing with asterisk/freePBX/allstar for a little over a year now and I am confident with it, but this one has me stumped now…And it’s tough to find documentation on this. or I am other thinking this and not able to wrap my head around it.

Thanks much for letting me ramble.

Jeff

···

Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net

Jeff - it’s funny that you mention this… I have broken down here: http://enhanced.github.io/2016/04/AllStar_AutoPatch how I got autopatch working with a SIP service, which is what the freepbx trunk should look like to ASL…

It’s funny that you mention this because I’m hoping to sit down next week and work through some of this just to get GVSIP working since that’s Googles new incantation of “SIP” and publish some documentation.

JJC

N0PKT

···

On Fri, Nov 16, 2018 at 12:05 PM Jeff Lehman kc8qch@gmail.com wrote:

So since I am on a roll today, here’s another to Chew on.

I have Incredible PBX setup on a Pi here. I use it (at least till Google shut off the GVSIP service today) for my home phone as well as Allstar and Hamshack Hotline (See hamshackhotline.com for more info).

I want to make one of my allstar nodes dial out through my PBX…Incredible PBX is based on FreePBX and Asterisk sooo…

I have calling from the PBX to ASL working fine. I can dial an extension from any of my phones and it connects to my ASL nodes.

I am stumped on From ASL to PBX. Here is what I have done…

I followed the Directions on this page: http://docs.allstarlink.org/drupal/node/133

So first I went into Rpt.conf and set those settings. I used "pstn-out"for the context in Rpt.conf for the autopatch. Next I went into extensions.conf and set the following in pstn-out:

[pstn-out]

exten => _XXX,1,Dial,SIP/pbx-autopatch/${Exten}

exten => _XXX,2, congestion

Note that XXX is for internal extensions. I use 3 digit extensions internally for phones. Next I went to sip.conf and set up the following stanza:

[pbx-autopatch]

type=peer

allowguest=yes

nat=yes

autocreatepeer=yes

insecure=port,invite

auth=md5

username=

password=nunya

host=

qualify=no

disallow=all

allow=ulaw

allowexternaldomains=yes

context=pstn-out

When I dial *61 + internal extension (203 for example) it waits a few seconds then gives me fast busy

When I watch on the asterisk CLI it tries to connect is not able to connect and at one point says “no such host: pbx-autopatch”

(I would paste the log but I am not home to be able to copy and paste)

Any initial thoughts? I’ve been playing with asterisk/freePBX/allstar for a little over a year now and I am confident with it, but this one has me stumped now…And it’s tough to find documentation on this. or I am other thinking this and not able to wrap my head around it.

Thanks much for letting me ramble.

Jeff


Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net


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Oh awesome I will take a look at it and see what you have.

Don’t bang your head on the GVSIP wall however. Sadly Google has stated that asterisk connections to GVSIP are a violation of TOS and has turned off GVSIP access to all but OBIHAI devices…

http://nerdvittles.com/?p=27274

I had 2 GVSIP trunks coming into the PBX and neither of them work today.

Jeff

···

Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net

Yeah I saw that, as luck has it I have an OBIHAI device and have read the article that you noted several times, just not had time to actually monkey with it lol.

···

On Fri, Nov 16, 2018 at 12:15 PM Jeff Lehman kc8qch@gmail.com wrote:

Oh awesome I will take a look at it and see what you have.

Don’t bang your head on the GVSIP wall however. Sadly Google has stated that asterisk connections to GVSIP are a violation of TOS and has turned off GVSIP access to all but OBIHAI devices…

http://nerdvittles.com/?p=27274

I had 2 GVSIP trunks coming into the PBX and neither of them work today.

Jeff

On Fri, Nov 16, 2018 at 2:12 PM JJC cummingsj@gmail.com wrote:

Jeff - it’s funny that you mention this… I have broken down here: http://enhanced.github.io/2016/04/AllStar_AutoPatch how I got autopatch working with a SIP service, which is what the freepbx trunk should look like to ASL…

It’s funny that you mention this because I’m hoping to sit down next week and work through some of this just to get GVSIP working since that’s Googles new incantation of “SIP” and publish some documentation.

JJC

N0PKT

On Fri, Nov 16, 2018 at 12:05 PM Jeff Lehman kc8qch@gmail.com wrote:

So since I am on a roll today, here’s another to Chew on.

I have Incredible PBX setup on a Pi here. I use it (at least till Google shut off the GVSIP service today) for my home phone as well as Allstar and Hamshack Hotline (See hamshackhotline.com for more info).

I want to make one of my allstar nodes dial out through my PBX…Incredible PBX is based on FreePBX and Asterisk sooo…

I have calling from the PBX to ASL working fine. I can dial an extension from any of my phones and it connects to my ASL nodes.

I am stumped on From ASL to PBX. Here is what I have done…

I followed the Directions on this page: http://docs.allstarlink.org/drupal/node/133

So first I went into Rpt.conf and set those settings. I used "pstn-out"for the context in Rpt.conf for the autopatch. Next I went into extensions.conf and set the following in pstn-out:

[pstn-out]

exten => _XXX,1,Dial,SIP/pbx-autopatch/${Exten}

exten => _XXX,2, congestion

Note that XXX is for internal extensions. I use 3 digit extensions internally for phones. Next I went to sip.conf and set up the following stanza:

[pbx-autopatch]

type=peer

allowguest=yes

nat=yes

autocreatepeer=yes

insecure=port,invite

auth=md5

username=

password=nunya

host=

qualify=no

disallow=all

allow=ulaw

allowexternaldomains=yes

context=pstn-out

When I dial *61 + internal extension (203 for example) it waits a few seconds then gives me fast busy

When I watch on the asterisk CLI it tries to connect is not able to connect and at one point says “no such host: pbx-autopatch”

(I would paste the log but I am not home to be able to copy and paste)

Any initial thoughts? I’ve been playing with asterisk/freePBX/allstar for a little over a year now and I am confident with it, but this one has me stumped now…And it’s tough to find documentation on this. or I am other thinking this and not able to wrap my head around it.

Thanks much for letting me ramble.

Jeff


Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net


App_rpt-users mailing list

App_rpt-users@lists.allstarlink.org

http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.


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To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.


Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net


App_rpt-users mailing list

App_rpt-users@lists.allstarlink.org

http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

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HA well there you go that works! Good Luck with that! :slight_smile:

···

On Fri, Nov 16, 2018 at 2:21 PM JJC cummingsj@gmail.com wrote:

Yeah I saw that, as luck has it I have an OBIHAI device and have read the article that you noted several times, just not had time to actually monkey with it lol.

On Fri, Nov 16, 2018 at 12:15 PM Jeff Lehman kc8qch@gmail.com wrote:

Oh awesome I will take a look at it and see what you have.

Don’t bang your head on the GVSIP wall however. Sadly Google has stated that asterisk connections to GVSIP are a violation of TOS and has turned off GVSIP access to all but OBIHAI devices…

http://nerdvittles.com/?p=27274

I had 2 GVSIP trunks coming into the PBX and neither of them work today.

Jeff

On Fri, Nov 16, 2018 at 2:12 PM JJC cummingsj@gmail.com wrote:

Jeff - it’s funny that you mention this… I have broken down here: http://enhanced.github.io/2016/04/AllStar_AutoPatch how I got autopatch working with a SIP service, which is what the freepbx trunk should look like to ASL…

It’s funny that you mention this because I’m hoping to sit down next week and work through some of this just to get GVSIP working since that’s Googles new incantation of “SIP” and publish some documentation.

JJC

N0PKT

On Fri, Nov 16, 2018 at 12:05 PM Jeff Lehman kc8qch@gmail.com wrote:

So since I am on a roll today, here’s another to Chew on.

I have Incredible PBX setup on a Pi here. I use it (at least till Google shut off the GVSIP service today) for my home phone as well as Allstar and Hamshack Hotline (See hamshackhotline.com for more info).

I want to make one of my allstar nodes dial out through my PBX…Incredible PBX is based on FreePBX and Asterisk sooo…

I have calling from the PBX to ASL working fine. I can dial an extension from any of my phones and it connects to my ASL nodes.

I am stumped on From ASL to PBX. Here is what I have done…

I followed the Directions on this page: http://docs.allstarlink.org/drupal/node/133

So first I went into Rpt.conf and set those settings. I used "pstn-out"for the context in Rpt.conf for the autopatch. Next I went into extensions.conf and set the following in pstn-out:

[pstn-out]

exten => _XXX,1,Dial,SIP/pbx-autopatch/${Exten}

exten => _XXX,2, congestion

Note that XXX is for internal extensions. I use 3 digit extensions internally for phones. Next I went to sip.conf and set up the following stanza:

[pbx-autopatch]

type=peer

allowguest=yes

nat=yes

autocreatepeer=yes

insecure=port,invite

auth=md5

username=

password=nunya

host=

qualify=no

disallow=all

allow=ulaw

allowexternaldomains=yes

context=pstn-out

When I dial *61 + internal extension (203 for example) it waits a few seconds then gives me fast busy

When I watch on the asterisk CLI it tries to connect is not able to connect and at one point says “no such host: pbx-autopatch”

(I would paste the log but I am not home to be able to copy and paste)

Any initial thoughts? I’ve been playing with asterisk/freePBX/allstar for a little over a year now and I am confident with it, but this one has me stumped now…And it’s tough to find documentation on this. or I am other thinking this and not able to wrap my head around it.

Thanks much for letting me ramble.

Jeff


Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net


App_rpt-users mailing list

App_rpt-users@lists.allstarlink.org

http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

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You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.


Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net


App_rpt-users mailing list

App_rpt-users@lists.allstarlink.org

http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users

To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the “Unsubscribe or edit options button”

You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.


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Jeff Lehman, KC8QCH
E-mail: kc8qch@gmail.com
http://kc8qch.dx.am
Hamshack Hotline: 4218

Webmaster
Hamilton County ARPSC
http://www.hamcoarpsc.org
E-mail: hamcoarpsc@gmail.com
Phone: 513-452-6480

Allstar 47374 Administrator
Project Manager, HHUL
World Wide Amateur Radio Guild
http://www.theguildglobal.org
E-Mail: kc8qch@theguildglobal.net