Does anyone know the setup so I can use VOIP Cheap as out bound all the how to's I have seen do not work....
 
VOIPCHeap
Submitted by longneck on Fri, 06/20/2008 - 3:45pm.
YOU can use my services if
Fri, 06/20/2008 - 6:09pm
YOU can use my services if you wish to, I just became a provider
I can give all the configuration
Allex
 
Allex,
Can you tell the
Fri, 06/20/2008 - 10:37pm
Allex,
Can you tell the community a bit about what you are doing?
Just a little background on your network? Are you reselling service on someone else's network?
If you are a facilities based provider we would like to hear about your network. Other ITSP's have come in and talked about their switches, gateways, proxies.
This information helps users make a choice. Sometimes a smaller company that serves one region well is a better option than a provider that spreads themselves too thin.
Good luck on your new venture....
Scott
--
Scott
aka "Skyking"
 
Hi Scott you Said
Fri, 06/20/2008 - 10:49pm
Hi Scott you Said everything
I was here to try to get a a solution for a costumer which has a ISA server configurations problems.
Well I am a resseler from one of the big ( I belive) US companies. They (we or I)have good call rates, unlimitted call plans to fixe numbers and cell phones, DIDs and everything else.The call quality is very good, and I advice everyone to join my Services.
Thanks to be so kind
I will be more them happy to help anyone
I am open for Questions
Best Regards
Allex
 
Thanks for the info
Fri, 06/20/2008 - 11:29pm
Thanks for the info Allex.
The reason I took a personal interest is there are a few folks here who operate or are in the process of building their own network.
Helping out here on trixbox is helping me learn trixbox and Asterisk. I build Service Provider Networks for a living, I am always interested in new network builds.
Other people would just say I am nosey!
Scott
--
Scott
aka "Skyking"
 
well I would say you are
Fri, 06/20/2008 - 11:34pm
well I would say you are curius
But I will be please if anyone here uses my services.
Let me know if you have costumers for me
Best Regards
Allex
 
....
Sat, 06/21/2008 - 1:21am
Well that was a nice side conversation Alex but you did not show anyone that you actually had the knowlege in the first place, if I knew I would be happy to help, I also currently run a large company but I do not try and sell my services everywhere.... The Asterix community will grow if we help each other not if we all try and charge for everything.
So does anyone have a setup which will work for voipcheap.com ???
Thanks..
 
Not a hijack
Sat, 06/21/2008 - 9:18am
You may have wait until somebody with specific knowledge of Voipcheap speaks up.
However, trunks we can help setup. Why don't you tell use what you have done, what is not working and what you are doing it on.
All the usual suspects, trix version, trunk config, network setup.
At some point we may ask for a log capture or a SIP debug. If we do please use the code tags if it is short or pastebin.ca if it is long.
I am not sure that folks realize what a pain it is to try and figure something out with 8000 lines of unrelated Asterbabble pasted in the middle of a thread.
Please read this thread. Following the spirit and not just the instructions will get you the quickest resolution to your problem.
Scott
--
Scott
aka "Skyking"
 
VoipCheap, Voipdiscount,
Sat, 06/21/2008 - 9:52am
VoipCheap, Voipdiscount, Voipfree call
They are all the same company I belive
The charge connections fee, that is why their prices are cheap.
Below is the configuration
allow=ulaw&alaw
authuser=yourusername
disallow=all
fromdomain=sipdiscount.com
fromuser=yourusername
host=sip.sipdiscount.com
insecure=very
nat=yes
qualify=yes
secret=yourpassword
sendrpid=yes
type=peer
username=yourusername
 
voipcheap etc
Sat, 06/21/2008 - 10:57am
They are all part of betamax. You can compare the rates here, and work out which 'trunk' is the best for the countries you want to call. http://www.voipproducts.eu/betamax-rates/
I have used voipcheap for quite a while - it seems fine
 
They are fine but they
Sat, 06/21/2008 - 11:03am
They are fine but they should satate the connection FEE!!
 
Thanks everyone
Sat, 06/21/2008 - 11:04am
Thanks I am playing around with this config now, stil not registering but assume it must be just getting things right like the register string etc... I am just putting this straight into the outbound route box and saving each time, still not there but guess it's a matter of time till I get it right...
by the way is the limit for voipcheap still 3 sessions ??
 
what is the connection fee
Sat, 06/21/2008 - 11:19am
what is the connection fee ?
as you point out I never saw one...
 
Connection fee is a rate
Sat, 06/21/2008 - 11:26am
Connection fee is a rate that they chearge as soon your call is connected
 
Yes but how much is it
Sat, 06/21/2008 - 11:32am
Yes but how much is it ?
Sorry I was not clear before, I cannot see on their site, at least it is not obvious...
 
That is never obvius, they
Sat, 06/21/2008 - 11:37am
That is never obvius, they dont say,
even your mobile provider does that. is a way to keep profit and lower the minute price rate,
 
The fee can be anything from
Sat, 06/21/2008 - 11:37am
The fee can be anything from 2 cents to 10
 
At a quick glance of your
Sat, 06/21/2008 - 1:31pm
At a quick glance of your configs that you posted I think some of your problem might be the order in which you have entered the information
You top 3 lines are"
allow=ulaw&alaw
authuser=yourusername
disallow=all
So basically if I recall right you are telling it to allow ulaw&alaw and than telling it to disallow all. It should be
disallow=all
allow-ulaw&alaw
and than you can enter the rest of your content. I hope this helps. If not you will have to find someone smarter than me.
 
I have in trunk settings
Sat, 06/21/2008 - 2:32pm
I have in trunk settings :
Outgoing settings:
Peer details:
username=yourusername
type=peer
secret=yourpassword
host=sip.voipcheap.com
incoming settings:
empty
Register string:
yourusername:yourpassword@sip.voipcheap.co.uk/yourusername
This works great on voipcheap.co.uk for outgoing calls, should work just as well on all the other Betamax services once you've changed the host (and user and pass of course). Infact I also use same settings on voipdiscount and smslisto without problems.
Antony
 
THanks that was very
Sat, 06/21/2008 - 3:56pm
THanks that was very clear,
It's odd the trunk is still not registering...
yet voipuser.org is....
I wonder what could be the matter
 
voipcheap.co.uk and
Sun, 06/22/2008 - 4:18am
voipcheap.co.uk and voipcheap.com are two different services (both Betamax but both have different tariffs and login), maybe something to do with you having host settings for one while using the other?
Antony
 
Is it real service?
Fri, 07/11/2008 - 7:35pm
I've found complaints from people who lost money or unable to make calls. Is it really good service?
Check this:
http://voipguides.blogspot.com/2008/02/betamax-voip-freezes-accou...
or this:
http://www.complaintsboard.com/complaints/wwwvoipcheapcom-c23798....
 
Ive been a very happy
Sat, 07/12/2008 - 6:35am
Ive been a very happy voipcheap customer for a couple of years now, I make most of my calls through them but have other lines just incase and for incoming calls, ive never had a time I couldnt make a call, never had a connection that was anything but perfect and never had any kind of billing problem.
I signed up to them fully expecting theyre may be the odd hiccup but that was tolerable as a home user and part of the trade of for very cheap or free calls, I have absolutely no complaints.
In regards to the two complaints above, one is very old (2005) the other I have seen before on this very forum when it was called out and the owner of yet another voip blog that just regurgitates stuff it found on the net had to backdown on his claims (iirc).
This is not to say they are perfect, I'm sure some, maybe even many had problems, more so in the past, but its a very cheap outgoing service, what have you got to loose, you will hopefully be more than satisfied, if you encounter a problem just fall back to your secondary provider, if completely dissatisfied ditch them and use another.
Enough with the sales pitch anyway, its not like I'm on commission.
PEER DETAILS
username=MYUSERNAME
type=peer
secret=MYSECRET
qualify=yes
nat=yes
insecure=very
host=sip.voipcheap.com
fromuser=MYPHONENUMBER
fromdomain=voipcheap.com
disallow=all
authuser=MYUSERNAME
allow=ulaw&alaw
REGISTER STRING
MYUSERNAME:MYSECRET@voipcheap.com
 
TDF for sure works for voice cheap
Sat, 07/12/2008 - 10:06am
How much do they pay you to say good things about them TDF???
 
voipcheap.com settings that work!!
Sun, 07/13/2008 - 8:31am
My voipcheap settings:
Peer Details:
allow=g729&g726&ulaw&alaw ;use the ones you want
authuser=username
disallow=all
fromdomain=sip.voipcheap.com
fromuser=username ;put your DID that you use for caller ID instead if you have registerd one
host=sip.voipcheap.com
insecure=very
nat=yes
qualify=yes
secret=password
type=peer
username=username
User Details:
(empty)
Register String:
username:password@sip.voipcheap.com/
That's it...it works for my voipbuster.com account as well (same company).
voipcheap.co.uk is more expensive in rates and offers less countries with free minutes.
voipcheap.com is better qua cost as well..offers loads more countries with free minutes.
I don't pay any connections fees...your charged for the first minute after the call is picked up..if it's a free distinations then it come from your free minutes.
It saves me and my family that are all over the world hooked up to my trixbox system lot's of money.I have not seen better rates than these anywhere else..
Good luck!!
Rene
 
Calls go out fine, except...
Fri, 11/14/2008 - 2:56pm
Caller ID continues to show as UNKNOWN. I've registered a DID with Gizmo5 and set that number up in my trunk, but callers still receive UNKNOWN. Below are my PEER details for the trunk:
username=MYUSERNAME
type=peer
secret=MYSECRET
qualify=yes
nat=yes
insecure=very
host=sip.voipcheap.com
fromuser=MY_GIZMO5_DID
fromdomain=sip.voipcheap.com
sendrpid=yes
disallow=all
authuser=MYUSERNAME
allow=ulaw&alaw
Furthermore, my register string is:
MYUSERNAME:MYSECRET@voipcheap.com
The calls go through and sound great, just have this Caller ID issue left to fix.
 
VOIPCHEAP ALL CIRCUITS ARE BUSY NOW
Wed, 02/03/2010 - 1:06pm
I see that some of you were trying to get voipcheap to work. I have tried many configs and no luck.
I am 100% positive about my network and SIP_NAT they are ok since other trunks work great.
I have trunk registered:
sip.voipcheap.com:5060 mstusa 105 Registered
voipcheapcom /mst 194.120.0.198 N 5060 OK (119 ms)
SIP TRUNK: OUTGOING PEER DETAILS
allow=ulaw&alaw
authuser=mst
disallow=all
fromdomain=sip.voipcheap.com
fromuser=mst
host=sip.voipcheap.com
insecure=very
nat=yes
qualify=yes
secret=password
sendrpid=yes
type=peer
username=mst
INCOMING empty
REGISTRATION STRING: user:password@sip.voipcheap.com/user
OUTBOUND ROUTE to that trunk with dial plan: 3|. - I have trying other plans like 001. or 011. with no luck
-- Executing [3001XXXXXXXXX@from-internal:1] Macro("SIP/2103-09cdb708", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2103-09cdb708", "AMPUSER=2103") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2103-09cdb708", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2103-09cdb708", "1|Set|REALCALLERIDNUM=2103") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2103-09cdb708", "AMPUSER=2103") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2103-09cdb708", "AMPUSERCIDNAME=USER") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2103-09cdb708", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2103-09cdb708", "AMPUSERCID=2103") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2103-09cdb708", "CALLERID(all)="USER" <2103>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/2103-09cdb708", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/2103-09cdb708", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/2103-09cdb708", "Using CallerID "USER" <2103>") in new stack
-- Executing [3001XXXXXXXXX@from-internal:2] Set("SIP/2103-09cdb708", "_NODEST=") in new stack
-- Executing [3001XXXXXXXXX@from-internal:3] Macro("SIP/2103-09cdb708", "record-enable|2103|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/2103-09cdb708", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/2103-09cdb708", "recordingcheck|20100203-140337|1265227417.211") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100203-140337|1265227417.211: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/2103-09cdb708", "") in new stack
-- Executing [3001XXXXXXXXX@from-internal:4] Macro("SIP/2103-09cdb708", "dialout-trunk|9|001XXXXXXXXX||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/2103-09cdb708", "DIAL_TRUNK=9") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2103-09cdb708", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2103-09cdb708", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/2103-09cdb708", "DIAL_NUMBER=001XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/2103-09cdb708", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/2103-09cdb708", "OUTBOUND_GROUP=OUT_9") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2103-09cdb708", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2103-09cdb708", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/2103-09cdb708", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/2103-09cdb708", "outbound-callerid|9") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2103-09cdb708", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2103-09cdb708", "0|Set|REALCALLERIDNUM=2103") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/2103-09cdb708", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/2103-09cdb708", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/2103-09cdb708", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/2103-09cdb708", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/2103-09cdb708", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/2103-09cdb708", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/2103-09cdb708", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/2103-09cdb708", "0|SetCallerPres|prohib_passed_screen") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/2103-09cdb708", "0|AGI|fixlocalprefix") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/2103-09cdb708", "OUTNUM=001XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/2103-09cdb708", "custom=SIP/voipcheapcom") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2103-09cdb708", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/2103-09cdb708", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2103-09cdb708", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2103-09cdb708", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2103-09cdb708", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/2103-09cdb708", "SIP/voipcheapcom/001XXXXXXXXX|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/2103-09cdb708", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/2103-09cdb708", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/2103-09cdb708", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 20) - failing through to other trunks") in new stack
-- Executing [3001XXXXXXXXX@from-internal:5] Macro("SIP/2103-09cdb708", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/2103-09cdb708", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/2103-09cdb708> Playing 'all-circuits-busy-now' (language 'en')
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/2103-09cdb708' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/2103-09cdb708'
DId anyone get this working? Thank You
 
my working voipcheap config
Thu, 02/18/2010 - 9:32am
I've noticed username in registered trunk is "mstusa" and it is different from "mst" in peer details. Also not clear what user:password is used in register string.
I have voicecheap working with no problem for couple of months. Here are my Outgoing peer details config (user:password are replaced with UUUU:PPPP)
type=peer username=UUUU secret=PPPP nat=no insecure=very host=sip.VoipCheap.com fromuser=UUUU fromdomain=sip.VoipCheap.com context=from-pstn
Incoming settings - empty
Register string:
UUUU:PPPP@sip.voipcheap.com
 
Thank You, I have already
Thu, 02/18/2010 - 10:18am
Thank You, I have already get this working for me. However, I was able to do only one free call outside of US and that's all. I have message from VoipCHeap that You have reached of maximum of free trial calls .........
Thank You
 
well, I guess it should be
Fri, 02/19/2010 - 8:27am
well, I guess it should be enough to test. Buy a small credit to test if further if you need it.
My system is completing 10-15 calls daily since last October through them and I have no problem so far.
 
Member Since:
2008-06-19