Asterisk SIP Trunking using extension instead of SIP username

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dcole13

534 posts

Ultimate Geek

#151339 21-Aug-2014 21:45

Hi,

I'm trying to hook my 2talk account up to my Asterisk server. I have followed this tutorial, but when I dial out it tries to register with "600@sip.2talk.co.nz", 600 is the extension number. Why is this happening and how can I fix it?

Thanks

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ajobbins

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#1113185 21-Aug-2014 22:50

This might be a better tutorial to follow

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sbiddle

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#1113216 22-Aug-2014 07:27

Those settings are very generic and aren't going to work with 99% of SIP providers. You need to be following what 2talk are telling you to use.

dcole13

534 posts

Ultimate Geek

#1113570 22-Aug-2014 17:06

ajobbins: This might be a better tutorial to follow

Hi,

Thanks for that. I am now able to dial in and out, but since I would like to have multiple SIP accounts how can I make it so SIP account 1 goes to, say extn 600 and SIP account 2 to extn 601?

Thanks

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ajobbins

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#1113575 22-Aug-2014 17:09

dcole13:
ajobbins: This might be a better tutorial to follow

Hi,

Thanks for that. I am now able to dial in and out, but since I would like to have multiple SIP accounts how can I make it so SIP account 1 goes to, say extn 600 and SIP account 2 to extn 601?

Thanks

Inbound routes and outbound routes

You may also want to consider IAX trunking, which should allow you have have alls calls routed down a single registration to 2talk, rather than per SIP account. Details here

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dcole13

534 posts

Ultimate Geek

#1113578 22-Aug-2014 17:13

ajobbins:
dcole13:
ajobbins: This might be a better tutorial to follow

Hi,

Thanks for that. I am now able to dial in and out, but since I would like to have multiple SIP accounts how can I make it so SIP account 1 goes to, say extn 600 and SIP account 2 to extn 601?

Thanks

Inbound routes and outbound routes

You may also want to consider IAX trunking, which should allow you have have alls calls routed down a single registration to 2talk, rather than per SIP account. Details here

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

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ajobbins

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#1113584 22-Aug-2014 17:26

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

No, and actually I should have said SIP Peering. You will need at least one registration per provider.

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dcole13

534 posts

Ultimate Geek

#1113588 22-Aug-2014 17:32

ajobbins:

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

No, and actually I should have said SIP Peering. You will need at least one registration per provider.

I have registered every SIP provider, and I can call out with their individual extensions, but when I call any one of the numbers it just uses the default "exten => s,1,Dial(SIP/600)" How can I make it so I can seperate the SIP providers to go to different extensions? Do I have to replace "s" with something else?

Thanks

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ajobbins

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#1113613 22-Aug-2014 18:12

dcole13:
ajobbins:

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

No, and actually I should have said SIP Peering. You will need at least one registration per provider.

I have registered every SIP provider, and I can call out with their individual extensions, but when I call any one of the numbers it just uses the default "exten => s,1,Dial(SIP/600)" How can I make it so I can seperate the SIP providers to go to different extensions? Do I have to replace "s" with something else?

Thanks

Why not use FreePBX or something to do your config? Much easier. YOu can then easily set each line as a DDI for an extension, or setup inbound routes for the numbers as necessary.

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sbiddle

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#1113695 22-Aug-2014 19:56

dcole13:
ajobbins:

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

No, and actually I should have said SIP Peering. You will need at least one registration per provider.

I have registered every SIP provider, and I can call out with their individual extensions, but when I call any one of the numbers it just uses the default "exten => s,1,Dial(SIP/600)" How can I make it so I can seperate the SIP providers to go to different extensions? Do I have to replace "s" with something else?

Thanks

I'll be brutally honest here...

If you need to answer questions like that you need to sit down and find 100 - 200 hours spare and learn how Asterisk works. It's no good explaining how to do this because you need to fully understand the concepts, particularly contexts, and more importantly the security implications.

If you're not willing to invest this time I'd recommend using a FreePBX based distro, you should be able to get a good basic understanding on how things work in 10-20 hours.

dcole13

534 posts

Ultimate Geek

#1113698 22-Aug-2014 20:00

sbiddle:
dcole13:
ajobbins:

Hi,

Not all of the VoIP accounts will be on 2talk, can I still do it this way?

Thanks

No, and actually I should have said SIP Peering. You will need at least one registration per provider.

I have registered every SIP provider, and I can call out with their individual extensions, but when I call any one of the numbers it just uses the default "exten => s,1,Dial(SIP/600)" How can I make it so I can seperate the SIP providers to go to different extensions? Do I have to replace "s" with something else?

Thanks

I'll be brutally honest here...

If you need to answer questions like that you need to sit down and find 100 - 200 hours spare and learn how Asterisk works. It's no good explaining how to do this because you need to fully understand the concepts, particularly contexts, and more importantly the security implications.

If you're not willing to invest this time I'd recommend using a FreePBX based distro, you should be able to get a good basic understanding on how things work in 10-20 hours.

I would rather learn about Asterisk and read for 100-200 hours than use something that does everything for me. Do you recommend any book/resource that would help me?

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chevrolux

4962 posts

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#1114884 24-Aug-2014 23:12

The Official FreePBX Distro is the best option for probably 99% of businesses that want to use an Asterisk-based VoIP solution.

There really is no point sitting down and writing dial plans for hours on end when what you are trying to achieve is a fairly standard thing but is rather tricky with no Asterisk knowledge.

FreePBX will allow you to create a very complex PABX system without needing to go anywhere near Asterisk dial plans - it is far more than just a GUI. When you use the official distro you will get CentOS, Asterisk, FreePBX, Fail2ban all ready to go in 10 minutes. Then you have the option to add commercial modules (spending $75 on the Endpoint Manager is the best money you will ever spend) if you require them and also get commercial support if you need it.

dcole13

534 posts

Ultimate Geek

#1114931 25-Aug-2014 08:43

chevrolux: The Official FreePBX Distro is the best option for probably 99% of businesses that want to use an Asterisk-based VoIP solution.

There really is no point sitting down and writing dial plans for hours on end when what you are trying to achieve is a fairly standard thing but is rather tricky with no Asterisk knowledge.

FreePBX will allow you to create a very complex PABX system without needing to go anywhere near Asterisk dial plans - it is far more than just a GUI. When you use the official distro you will get CentOS, Asterisk, FreePBX, Fail2ban all ready to go in 10 minutes. Then you have the option to add commercial modules (spending $75 on the Endpoint Manager is the best money you will ever spend) if you require them and also get commercial support if you need it.

I corrupted my asterisk and debian install so I might just try that out. Looks a lot easier.

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dcole13

534 posts

Ultimate Geek

#1115458 25-Aug-2014 19:54

I'll just post here because I don't think people will like a new topic, but when I install FreePBX it seems that it doesn't install Asterisk, because when I boot it, and log in nothing pops up, so I tried to run ./start_asterisk start and all it did was say that /usr/sbin/safe_asterisk didn't exist. I'm installing with Serva Netboot if it matters.

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chevrolux

4962 posts

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#1115467 25-Aug-2014 20:37

Did you get the official distro from freepbx.org?

I've always installed from either USB or ISO and it literally takes 10 minutes (plus about half an hour for module updates lol) and it works every single time. You should be able to browse to the admin panel and start configuring stuff.

dcole13

534 posts

Ultimate Geek

#1115472 25-Aug-2014 20:40

chevrolux: Did you get the official distro from freepbx.org?

I've always installed from either USB or ISO and it literally takes 10 minutes (plus about half an hour for module updates lol) and it works every single time. You should be able to browse to the admin panel and start configuring stuff.

The PC I'm using won't boot from CD or USB so I had to use netboot.

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