Asterisk configuration

[FONGO_steve]

Going to look into this further today and see what's going on for everyone

[teamer]

The change in sip_custom.conf made it working for me. I just added a useragent="useragent" row and restarted.

Thanks

[FONGO_kris]

Our backend does not authenticate via useragent and DOES support Asterisk... Weird.

If that works, good. But we're still looking into the entire issue.

[BelMarduk]

Hi all,

I have been running "PBX in a Flash" asterisk distro. Everything has been working fine for the past year until this issue happened.
Under "Free PBX" -> Asterisk SIP settings I added the

useragent=LinksysPAP2TNA

under "Other SIP Settings", since with Free PBX you are not supposed to edit the sip.conf or sip_custom.conf manually as it will get overwritten by FreePBX.

But now when I make a call out from this line, it connects to the other end, but I cannot hear anything. The other party on the other side does hear me, but I cannot hear them.

Could anyone help me? Perhaps the proper trunk settings for freephoneline.ca. This is what I have:

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Trunk Name: freephoneline.ca
[u]PEER Details[/u]
username=1416XXXXXXX 
type=peer
sendrpid=yes
secret=ITISASECRET
nat=yes
insecure=very
qualify=no&yes
host=voip.freephoneline.ca
fromuser=1416XXXXXXX 
fromdomain=voip.freephoneline.ca
context=from-freephoneline
disallow=h263&h263p
dtmfmode=auto
allow=G729a&ulaw&g729

Code: Select all

USER Context: 1416XXXXXXX 
[u]USER Details[/u]
type=user
fromdomain=voip.freephoneline.ca
fromuser=1416XXXXXXX
secret=ITISASECRET

Code: Select all

Register String:
1416XXXXXXX:ITISASECRET@voip.freephoneline.ca/1416XXXXXXX

[nicolas_dh]

I have been running "PBX in a Flash" asterisk distro. Everything has been working fine for the past year until this issue happened.
...
with Free PBX you are not supposed to edit the sip.conf or sip_custom.conf manually as it will get overwritten by FreePBX.
...
But now when I make a call out from this line, it connects to the other end, but I cannot hear anything. The other party on the other side does hear me, but I cannot hear them.

Hello BelMarduk,

According to this website for "PBX in a Flash" (PIAF): http://knol.google.com/k/pbx-in-a-flash ... )Way_Audio
within the section: Getting Rid of One-Way Audio (that you seem to be experiencing now)
the author propose to adjust sip_custom.conf using "nano -w" ?!?!?(will it be overwritten by FreePBX)
to specify your externip "static" IP see http://www.voip-info.org/tiki-index.php ... P+externip

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externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0      (NOTE: The first 3 octets need to match your private IP addresses!)

Note: If your public IP is not static, the author recommends some script and cron settings.


You may also want to have the option canreinvite set to no
check the options at: http://www.voip-info.org/wiki/view/Aste ... anreinvite

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canreinvite=no

You should also ensure your firewall forwards SIP & RTP (& IAX2) ports to your "PBX in a Flash" see
http://pbxinaflash.org/Tutorials section Port Forwarding from your Router to PIAF

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UDP 10000-20000 - RTP
UDP 5004-5082 - SIP
UDP 4569 - IAX2

As your setup was working before this issue happened
(FPL'Sippy denied-sip-Invite for specific useragent since 2009-12-14),
maybe adding the "useragent=LinksysPAP2TNA" in "Other SIP Settings", broke your configuration?!?
Otherwise this "issue" could be broader than just the "useragent" denied-sip-invite ?

[FONGO_kris]

Not to say I completely condone the use of our service with Asterisk since it's hard to snuff out the businesses; here is something that will put even more emphasis on what Steve and I have been stressing, which is our SIP Server does not block Asterisk

This system, though actually quite simple, is a quick synopsis of our servers and how they terminate.

http://www.portaone.com/resources/featu ... terisk.pdf... and don't forget to read http://www.portaone.com/news/news_detail.php?ID=1436

[nicolas_dh]

Thanks for allowing(or condoning) the use of FPL service with Asterisk.
It is great, even if we have to set:
useragent=LinksysPAP2TNA
FPL shows openness allowing the home use of Asterisk.
I believe everyone will eventually migrate to FPL,
many without asterisk... but reassured by friends that are using home PBX(s).

[repherb]

Hi everyone!

Simple but stupid question:
I just bought a Digium TDM22B 2FXO/2FXS card and want to get PBX In A Flash working with FPL as an inbound/outbound trunk because i want to play with VoIP as a hobby that might lead me into a new career someday.
I also have a Bell PSTN line i want to connect to 1 FXO port as an inbound/outbound fallover trunk in case of an internet outage/FPL failure (not saying FPL would go down themselves, i just love the company, their service, and their vision for the future!)

I have no bloody idea how to do it, and i am stumbling at every turn. No one out there has any information on how to do this, but apparently it can be done.
So far, i cannot call inbound to the box with this configuration:
[PEER]
username=1289639XXXX
secret=SECRET
allow=ulaw&g729
bindport=5060
canredirect=no
defaultexpiry=3600
disallow=all
dtmfmode=rfc2833
fromuser=1289639XXXX
host=voip.freephoneline.ca
insecure=very
maxexpirey=3600
nat=no
qualify=no
type=friend

[USER]
allow=ulaw&g729
canredirect=no
context=from-trunk
fromdomain=voip.freephoneline.ca
fromuser=1289639XXXX
type=user
username=1289639XXXX
secret=SECRET

registration=1289639XXXX:SECRET@voip.freephoneline.ca/1289639XXXX

so with that said, how would i specify the outbound trunk to route calls to g2 or g3 (my FXO ports) and allow my g0 or g1 (my FXS ports) to ring inbound as extensions???

thanks in advace!

[dudu_georgescu]

Anybody using this with trixbox? I am able to send outbound calls, but have trouble with inbound calls ... Basically, the inbound calls reach the server, however instead of connecting to my inbound route, they end up playing the goodby tune (default) and then the call gets hung up. Any ideas?

Btw, I have a "catch all inbound" good for all incomming calls; and when dialing 7777 which simulates incomming calls, it gets triggered real easy which makes this stuff even more confusing ...

[koreyb]

Anybody using this with trixbox? I am able to send outbound calls, but have trouble with inbound calls ... Basically, the inbound calls reach the server, however instead of connecting to my inbound route, they end up playing the goodby tune (default) and then the call gets hung up. Any ideas?

Btw, I have a "catch all inbound" good for all incomming calls; and when dialing 7777 which simulates incomming calls, it gets triggered real easy which makes this stuff even more confusing ...

I use trixbox and have no issues with it....

[dudu_georgescu]

Another issue: no matter to what I set the dtmf mode, digits are not understood on the incomming route. The outbound seems not to have any issue, the inbound is the problem ...

[evilmonkey]

Hi good people,

I'm still having the 'account number not valid' problem. My details are below, any help is really appreciated. I'm running freepbx 2.6.x with asterisk 1.4.26.3.

TRUNK DESCRIPTION: freephoneline
OUTBOUND CALLER ID: 1647xxxx

Trunk name: freephoneline
PEER DETAILS:

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useragent=LinksysPAP2TNA
host=voip.freephoneline.ca
context=from-pstn
fromuser=1647xxxx
username=1647xxxx
secret=xxxx
type=peer
insecure=very

USER CONTEXT: 1647xxxxx
USER DETAILS:

Code: Select all

type=user
fromdomain=voip.freephoneline.ca
fromuser=1647xxxx
username=1647xxxx
secret=xxxx

registration string: 1647xxxx:xxxx@voip.freephoneline.ca/1647xxxx

sip_custom.conf

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[general]
useragent=LinksysPAP2TNA

I know that the useragent line probably shouldn't be in the PEER details, but I had the same result with and without it. The line is in my sip_custom.conf.

Thanks!

[evilmonkey]

Okay, I solved my problem. For those for whom editing sip_custom.conf doesn't work, open your sip.conf file and see what files it includes. For me, the problem was that sip_custom.conf was included after the general context has been defined. If I want to define something inside the general context, I have to add it to sip_general_custom.conf instead of sip_custom.conf. Adding useragent=pap2 (no need to add [general] to the top of the file) did the trick for me.

Thanks again to FPL for condoning asterisk, if only unofficially. I only ask that the current setup is not altered in the future to make using asterisk more difficult. As much as there is a need to keep out those who abuse the system, there is no need to penalize those of us who use it for legitimate reasons.

[FONGO_steve]

Thanks again to FPL for condoning asterisk, if only unofficially. I only ask that the current setup is not altered in the future to make using asterisk more difficult. As much as there is a need to keep out those who abuse the system, there is no need to penalize those of us who use it for legitimate reasons.

Most people don't realize the incredibly large amount of people who sign up purely to abuse it. Free phone numbers are nearly impossible to come by, and can be used for any number of illegal things. Unfortunately there comes a point where from a legal standpoint we must protect ourselves as best as we can from unauthorized use.
However, there are many discussions going around of soft PBX and business offerings in the future. Nothing official yet, but we want to be able to accommodate everyone, while keeping a close eye for illegal use and abuse.

[evilmonkey]

Sounds fair Steve. Sucks that a few bad apples screw this up for everyone.

[FONGO_steve]

Sounds fair Steve. Sucks that a few bad apples screw this up for everyone.

I know we're looking into actually offering some soft PBX options in the future though! But like I say, nothing official, and definitely no time frames available yet, I know there are a lot of other huge tasks that have priority for 2010!

[sukasem]

Hi,

Anyone has IVR menu working? FPL trunk won't recognise DTMF tone when I press menu option.

Cheers,
Sukasem

[evilmonkey]

Hi,

Anyone has IVR menu working? FPL trunk won't recognise DTMF tone when I press menu option.

Cheers,
Sukasem

Add dtmfmode=info in trunk details and peer details for the FPL trunk. That did the trick for me.

[sdertsr6]

great info here...thanx for posting..

[ham316]

I keep getting password is not valid message. What password am I supposed to use?