[larryalk]

Originally Posted by mberlant

inept,

Your other thread talks about this situation directly. Kindly return there to continue your quest to get your SPA3000 working with your Asterisk.

Before you can run (send inbound PSTN calls to an IVR for capture into various mailboxes) you need to learn to walk (implement an "s" stanza in your extensions.conf context).

I'd like to view the "other thread" but I'm not sure where it is. Can you give me a pointer to it?

Larry

[NghtShd]

Originally Posted by muppetmaster

Posted this on another site, though it might be useful here if not already embedded somewhere in a thread:

Sipura 3000 -> PSTN Config for Incoming

- Add a dialplan (ie - Dial Plan 1/Dial Plan 2) with a SIP URL destination as follows:

 S0<:1000@192.168.1.50> (where the IP address is that of your * instance, also ensure the extension is properly accessible in you extensions.conf)

How would one access the caller ID info in extensions.conf using this? If there a way to append the number calling to the extension? Alternatively, can the number be accessed in asterisk.

With SIP debug on I see asterisk knows the caller's number such as below:

From: Cell Phone   GA <sip:678XXXXXXX@192.168.0.2>;tag=1eb5f2e1ebc897d7o1

But I don't know how to get to that info in my asterisk incoming pstn dial plan.

[NghtShd]

Does anyone know of a list of macros available in Asterisk? I found a work-around for the problem from my previous post by using the ${CALLERID} macro (which I didn't find documented, I just guessed it). As great as the Asterisk PDF book is, an appendix with this kind of info would make it even greaterisher.

I still would rather have the linksys insert the calling number into the extension if possible, but if it's not I think I can do what I want with this.

[jperyy999]

I got the spa3000 basically working with Asterisk, but for some reason it takes two asterisk Trunks (one makes it work for incoming and one makes it work for outgoing). What can I fix so it is just one Trunk?

The only difference I see in the two definitions in sip_additional.conf ( [7001] vs [spa3000] ) is
host=192.168.1.201
vs
host=dynamic

What I have:

I downloaded and clean installed the latest trixbox a couple days ago.

Asterisk server is 192.168.1.200
spa3000 is 192.168.1.201

spa3000 "PSTN Line" has:
Proxy and Registration
Proxy: 192.168.1.200 Use Outbound Proxy: no
Outbound Proxy: (blank) Use OB Proxy In Dialog: yes
Register: no Make Call Without Reg: yes
Register Expires: 3600 Ans Call Without Reg: yes
Use DNS SRV: no DNS SRV Auto Prefix: no
Proxy Fallback Intvl: 3600 Proxy Redundancy Method: Normal
Subscriber Information
Display Name: PSTN1 User ID: spa3000
Password:***** Use Auth ID: no
Auth ID: (blank)
VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID: spa3000 VoIP User 1 DP: 1
VoIP User 1 Password: ******


Asterisk:
sip_additional.conf:

[7001]
username=spa3000
type=peer
secret=2222
port=5061
nat=no
insecure=very
host=192.168.1.201
context=from-pstn
canreinvite=no

[spa3000]
username=spa3000
type=peer
secret=2222
port=5061
nat=no
insecure=very
host=dynamic
context=from-pstn
canreinvite=no

[cometrider]

Originally Posted by jperyy999

I got the spa3000 basically working with Asterisk, but for some reason it takes two asterisk Trunks (one makes it work for incoming and one makes it work for outgoing). What can I fix so it is just one Trunk?

Of all the pages of instructions I've found on the web for using an SPA3000 with a Trixbox/FreePBX setup (including ones written by so-called "experts"), this one is the only one that actually worked for me. I think if you follow the instructions here, you will be okay, and using only a single trunk:

Setup Sipura 3000 with FreePBX - AussieVoIP

Of course, YMMV so save your existing settings "just in case" - better to have two trunks that work as opposed to one that doesn't.

[poi9m]

Asterisk and 2 voice gateways LinkSys SPA3000.
In options of one - pstn-spa3k, second pstn-spa3k.

One work perfect:
exten => _99XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60) ;v gorod
exten => _99XXXXXXX,2,Congestion


When I want to do a outgoing call thrue one and then other:
Algoritm good ?

exten => _99XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60)
exten => _99XXXXXXX,2,Goto(pstn-spa3k-${DIALSTATUS},1)
exten => _99XXXXXXX,3,Hangup
exten => _99XXXXXXX-NOANSWER,1,Dial(SIP/${EXTEN}@pstn-spa3k2,60,TtWw)
exten => _99XXXXXXX-BUSY,1,Dial(SIP/${EXTEN}@pstn-spa3k2,60,TtWw)

Interest 2-nd string, is it correct
Thanks !

[Vladje]

I'm trying to setup a spa3102, but the pstn-to-voip gateway isn't yet working like it should. Maybe someone can help me?

The gateway is setup to proceed a call to my trixbox with the ip 192.168.50.5 with a dailplan. It seems like the spa isn't proceeding the call, it isn't shown in the statuspage and this is what the log/debug sais when i'm calling it with a pstn phone:

06-28-2007 01:53:34 Local2.Debug 192.168.50.6 AUD:Stop PSTN Tone
06-28-2007 01:53:34 Local2.Debug 192.168.50.6 AUD:Stop PSTN Tone
06-28-2007 01:53:29 Local3.Debug 192.168.50.6 FXO:CNDD Name= Phone=45


The weird thing about is, that when i set the spa to proceed the calls to line 1, the status page shows that it is proceeding to the line1, just when i change this to the trixbox, then it does no longer work...

I really think I tried everything, so hope someone can help me.

[pcman800]

hello to all of you who knows (I don't know how to post here) VoIP well, anyway, I am new in VoIP, I just wanna ask a help from you. I have an AXON Server and Express Talk that installed to my pc and I have a SPA3000 also bought from US with my employer. He want the VoIP can call anywhrere without limit, means can call vice versa wether you are using softphone, ATA phone and PC. I dont know also what is dialingplan and what it the meaning of asterisk like this one =>([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>) and also this one =>(<S0:802>). I just got this when I configure it to the web wizard. Hoping a help from you. thnx



erwin of Philippines (pcman800@yahoo.com, pcman800@voxilla.com)

[pcman800]

hello phoneboy

I want your help about this VoIP phone and AXON server. I dont know how to configure this. Im just new about this. Could you please help me. Im very sick and tired configuring all of this (about 3 weeks of configuring) but still now i cant get it. My employer want to make a call for international and for local without the PSTN line. Although he have PSTN line one in her office. The model of VoIP phone is SPA3000. I have also installed Express talk to my pc so that I can call also to hongkong. Regarding the network it is ok. I have no idea also about dial plan, i don't what is x, *, |, (, ), <, >, ., :, S, 0, @, gw0, [, ], all of this stand for. My location, Philippines. Can possible can call also to cellphone?


Need your help very much



thanks,




erwin of Philippines (pcman800@yahoo.com)

[Evgen]

Can SPA3102's FXO-port pass CallerID from PSTN line to Asterisk without "off hook" to allow Asterisk ifself to take decision whether to answer the call or not?