VOIP / Asterisk / stuff

growse · 13 Mar 2007 at 20:26

Ok, so I want to leap on the VOIP bandwagon fairly cheaply. I've installed Asterisk on my gentoo server, and I'm about to signup with sipgate to get my number. I have a few questions:

1) Can I get a free number through sipgate and use them for incoming, and also get free outgoing UK landline calls through a different provider? Any provider recommendations for free UK landline calls?

2) Any good softphone options for linux or windows before I get a hardware sip phone?

3) Can my cisco 837 assist in any way with QoS?

4) Am I right in thinking that a dedicated WAN IP is best? I have one spare..

5) I presume Asterisk handles voicemail for me? I also presume this can do funky things like email an mp3 of the voicemail when I get it?

6) Any general asterisk setup guides?

Thanks

tolien · 13 Mar 2007 at 21:16

growse said:
2) Any good softphone options for linux or windows before I get a hardware sip phone?

X-lite was nice, the last I used it.

3) Can my cisco 837 assist in any way with QoS?

Probably not. Fair queueing was enough when I used my 837 to give me a decent call while I was uploading.

4) Am I right in thinking that a dedicated WAN IP is best? I have one spare..

No-NAT's the best option...SIP with NAT is potentially messy.

Pass on the rest, I never spent enough time looking at *.

Andri · 13 Mar 2007 at 23:08

theres a "wizard" version of asterisk build on centos which compiles itself and sets itself up.

Trixbox, it used to be called Asterisk@Home. its not too bad, it does its job. to make things easyer you can install the FreePBX web interface so you don't have to edit the config files through the cli.

you could probably set a badwidth limit and set is as priority for sip/iax if you go that route. although i'm not sure your router would be able to do that.

i'll see if i can dig out a good resource site for ya.

growse · 14 Mar 2007 at 10:49

Andri said:
theres a "wizard" version of asterisk build on centos which compiles itself and sets itself up.

Trixbox, it used to be called Asterisk@Home. its not too bad, it does its job. to make things easyer you can install the FreePBX web interface so you don't have to edit the config files through the cli.

you could probably set a badwidth limit and set is as priority for sip/iax if you go that route. although i'm not sure your router would be able to do that.

i'll see if i can dig out a good resource site for ya.

Thanks - I'll look at freepbx. I've compiled asterisk on gentoo last night, but havn't had a chance to play with it. Although my server now isn't letting me ssh in, so I'll have to figure out why that is....

Topgun · 14 Mar 2007 at 11:34

A good voip ata and a voxalot account is the way to go. Asterisk is overkill imo.

Andri · 14 Mar 2007 at 12:04

Topgun said:
A good voip ata and a voxalot account is the way to go. Asterisk is overkill imo.

true but its free. and interesting to configure. a nice way to learn the more advanced side of voip. unless of course you do just need a turn on and ready to go package. in which case Draytek sell VOIP adsl routers which you can connect sip phones to and set up and external sip account.

Topgun · 14 Mar 2007 at 12:38

Andri said:
true but its free. and interesting to configure. a nice way to learn the more advanced side of voip. unless of course you do just need a turn on and ready to go package. in which case Draytek sell VOIP adsl routers which you can connect sip phones to and set up and external sip account.

Not exactly free because you will still need need an fxo/fxs device of some kind to use with asterisks. Have a look at this video to see what's involved.
http://youtube.com/watch?v=UP9b_FEZuUE

Topgun · 14 Mar 2007 at 12:48

1)To use sipgate for incomming and another provider for outgoing, you could use asterisks or just get a voxalot account and manage your providers from there.

3) A hardware sip device is definitely the way to go. The linksys PAP2 is very popular and you can plug any regular phones into it.

4) A dedicated IP is not necessary but could make things a little easier if you are not happy about forwarding ports. I don't use a static or dedicated IP with my voip service and it is fine.

5) Sipgate can do voicemail to mp3 email already

Personally I think voip.co.uk is better than sipgate. You really only need asterisks if you want to do stuff like music on hold, "press 1 to speak to sales" etc etc.

dangermouse · 14 Mar 2007 at 20:46

growse: Your asterisk compile on gentoo, what use flags are you using? I've been trying to get it to work much in the same way as you, Asterisk with SipGate. I've run into problems with asterisk though.

growse · 15 Mar 2007 at 00:09

dangermouse said:
growse: Your asterisk compile on gentoo, what use flags are you using? I've been trying to get it to work much in the same way as you, Asterisk with SipGate. I've run into problems with asterisk though.

[ebuild R ] net-misc/asterisk-1.2.13 USE="mysql ssl -alsa -bri -curl -debug -doc -genericjb -gtk -h323 -hardened -lowmem -mmx -nosamples -odbc -osp -postgres -pri -speex -sqlite -ukcid -zaptel" 0 kB

growse · 15 Mar 2007 at 16:33

I'm essentially tearing my hair out over this one. I've set up an account with sipgate, and asterisk refuses to connect to it. I've forwarded an entire ip address to the box that asterisk is running on, turned off the firewall, and still nothing. What the hell is going on?

Code:

Mar 15 16:20:35 NOTICE[8295] cdr.c: CDR simple logging enabled.
Mar 15 16:20:35 NOTICE[8309] chan_sip.c:    -- Re-registration for  [email protected]
Mar 15 16:20:55 NOTICE[8309] chan_sip.c:    -- Registration for '[email protected]' timed out, trying again (Attempt #1)
Mar 15 16:21:15 NOTICE[8309] chan_sip.c:    -- Registration for '[email protected]' timed out, trying again (Attempt #2)
Mar 15 16:21:35 NOTICE[8309] chan_sip.c:    -- Registration for '[email protected]' timed out, trying again (Attempt #3)
Mar 15 16:21:55 NOTICE[8309] chan_sip.c:    -- Registration for '[email protected]' timed out, trying again (Attempt #4)

My sip.conf looks like this:

Code:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

register => 8101384:[email protected]/8101384
externip = 88.96.138.221 
localnet=192.168.0.0/255.255.255.0

[SIPGATENUM]
type=peer
username=8101384
fromuser=8101384
secret=BIGFATPWD
context=default
host=sipgate.co.uk
fromdomain=sipgate.co.uk
insecure=very
caninvite=no
canreinvite=no
nat=yes
disallow=all
allow=ulaw
allow=alaw

[sipgate_de_in]
type=peer
fromdomain=sipgate.de
host=sipgate.co.uk
context=incoming
diallow=all
allow=ulaw
allow=alaw
nat=yes