provided by Star2Billing S.L.

Please can anyone help me in figuring out why I kept getting the number you dialed is unavailable...pls check the number and dial again.
All other trunks work just fine but couldn't figure out the settings with carriexchange.
created a SIP trunk
created SIP Client with my trunk ID
added carrie context to from-internal....local calls get busy signal only allowing international calling...Any quick reference will highly be appreciated. I tried all the steps given by carriexchange support did not help.

Ok.. First you have to make sure you are removing 011 from the dialed number (within A2B trunk config), that is if you use 011 on your rate cards. Next you have to append your trunk ID to the dialed number (this is also done in A2Billing Trunk Config). Please information below:

A2Billing Trunk Config
Label: Carriex
Add Prefix: Carriex Trunk ID
Remove Prefix: 011
Provider Tech: SIP
Provider IP: Inmy case carriex (this is the context name in Asterisk)
Additional Parameter: BLANK
Failover Trunk: Up to you

Asterisk Trunk Setting
Outbound Caller ID: Your Choice
Never Override: Not Checked
Max Cahnnels: Blank
Dial Rules: Blank
Outbound Dial Prefix: Blank - this is done in a2billing
Trunk Name: Carriex
Peer Details
allow=g729
canreinvite=no
context=from-trunk
disallow=all
fromuser=xxxx
host=64.71.145.237
insecure=very
secret=xxxx
type=friend
username=xxxx

Reg String: xxxxx:[email protected]

Make sure you have your carriexchange trunk configured properly and pointed to the IP of your Asterisk Box

How would you rate the quality of various trunks provided by users in Carriex? Poor Good or Excellent

Please refer to the CarrieX Provider Ratings by Zone
CarrieX Provider Ratings by Zone

Many of your CarrieX sub-provider questions will be answered here!

Problem was solved...had the right codec g729 and things went smooth.
There is no config for trixbox but i managed to get it to work with the following...Thanks for the reply.

allow=g729&g723
canreinvite=no
context=from-internal
fromuser=XXXX
host=64.71.145.237
insecure=very
secret=XXXXX
type=peer
username=XXXXX

my Dial rules looks like this:

XXXXX+011|Z.

I'm still unable to get through with CarrieXchange and trixbox. Can someone please help? All i got is All circuits are busy now, please try your call again later.

Could someone please show me what exactly you have for your trunk setting and outbound setting on trixbox and also on your carriexchange?



TRUNK Setting on A2Billing

VOIP-PROVIDER carriexchange
LABEL carriexchange
ADD PREFIX xxxxxx (my trunk id)
REMOVE PREFIX 011
PROVIDER TECH SIP
PROVIDER IP PROVIDER IP


TRUNK Setting on TRIXBOX
Outbound Caller ID xxx-xxx-xxxx (my own caller id)
Maximum channels <blank>

Outgoing Dial Rules
Dial rules xxxxxx+011|Z.

Outbound Dial Prefix: Blank - this is done in a2billing
Trunk Name: Carriex
Peer Details
allow=g729
canreinvite=no
context=from-trunk (I tried both from-trunk and from-internal)
disallow=all
fromuser=xxxx
host=64.71.145.237
insecure=very
secret=xxxx
type=friend
username=xxxx

Reg String: xxxxx:[email protected]


HERE IS THE LOG WHEN I TRY TO MAKE A CALL: (I replaced the trunk ID with xxxxxx)

== Manager 'admin' logged on from 127.0.0.1
-- Executing [19135682215@from-internal:1] Macro("SIP/204-b760b580", "dialout-trunk|7|19135682215||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/204-b760b580", "DIAL_TRUNK=7") in new stack
-- Executing [s@macro-dialout-trunk:2] Set("SIP/204-b760b580", "DIAL_NUMBER=19135682215") in new stack
-- Executing [s@macro-dialout-trunk:3] Set("SIP/204-b760b580", "ROUTE_PASSWD=") in new stack
-- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/204-b760b580", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing [s@macro-dialout-trunk:6] GotoIf("SIP/204-b760b580", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("SIP/204-b760b580", "_NODEST=") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("SIP/204-b760b580", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:9] Set("SIP/204-b760b580", "GROUP()=OUT_7") in new stack
-- Executing [s@macro-dialout-trunk:10] Macro("SIP/204-b760b580", "user-callerid|SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/204-b760b580", "user-callerid: device 204") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/204-b760b580", "AMPUSER=204") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/204-b760b580", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] GotoIf("SIP/204-b760b580", "0?start") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/204-b760b580", "REALCALLERIDNUM=204") in new stack
-- Executing [s@macro-user-callerid:6] NoOp("SIP/204-b760b580", "REALCALLERIDNUM is 204") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/204-b760b580", "AMPUSER=204") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/204-b760b580", "AMPUSERCIDNAME=Desktop X-Ten") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/204-b760b580", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/204-b760b580", "AMPUSERCID=204") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/204-b760b580", "CALLERID(all)="Desktop X-Ten" <204>") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/204-b760b580", "REALCALLERIDNUM=204") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/204-b760b580", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/204-b760b580", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/204-b760b580", "Using CallerID "Desktop X-Ten" <204>") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/204-b760b580", "record-enable|204|OUT") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/204-b760b580", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/204-b760b580", "recordingcheck|20080119-185431|1200786871.57") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080119-185431|1200786871.57: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/204-b760b580", "No recording needed") in new stack
-- Executing [s@macro-dialout-trunk:12] GotoIf("SIP/204-b760b580", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/204-b760b580", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:14] Macro("SIP/204-b760b580", "outbound-callerid|7") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/204-b760b580", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/204-b760b580", "REALCALLERIDNUM is 204") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/204-b760b580", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/204-b760b580", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/204-b760b580", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/204-b760b580", "TRUNKOUTCID=800-733-5830") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/204-b760b580", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/204-b760b580", "0?usercid") in new stack
-- Executing [s@macro-outbound-callerid:17] Set("SIP/204-b760b580", "CALLERID(all)=800-733-5830") in new stack
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/204-b760b580", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/204-b760b580", "CallerID set to "" <8007335830>") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/204-b760b580", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] AGI("SIP/204-b760b580", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern xxxxxx+011|Z.
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:18] Set("SIP/204-b760b580", "OUTNUM=xxxxxx19135682216") in new stack
-- Executing [s@macro-dialout-trunk:19] Set("SIP/204-b760b580", "custom=SIP/carriexchange") in new stack
-- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/204-b760b580", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,22)
-- Executing [s@macro-dialout-trunk:22] Macro("SIP/204-b760b580", "dialout-trunk-predial-hook") in new stack
-- Executing [s@macro-dialout-trunk:23] GotoIf("SIP/204-b760b580", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/204-b760b580", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:25] Dial("SIP/204-b760b580", "SIP/carriexchange/xxxxxx19135682216|300|") in new stack
-- Couldn't call carriexchange/xxxxxx19135682216
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dialout-trunk:26] Goto("SIP/204-b760b580", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/204-b760b580", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/204-b760b580", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing [19135682216@from-internal:2] Macro("SIP/204-b760b580", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/204-b760b580", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/204-b760b580> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/204-b760b580", "pls-try-call-later|noanswer") in new stack
-- <SIP/204-b760b580> Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/204-b760b580' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/204-b760b580'

I'm sure Seth will be around shortly with the definitive answer, but in the meantime my guess is:
There's a problem with your [carriexchange] peer definition in sip.conf. Your post mentions carriexchange, Carriex and an IP address. You need to make sure you refer to a trunk with its exact (case-sensitive) name in all instances.
I'm hoping you are aware your call is not passing through A2B at all. Of course this might be intentional and if so I'm very pleased as you've evidently been reading the forums; testing carriers with Asterisk alone before involving A2B is a smart move.
edited to add:
I would heartily recommend you swapping your insecure=very for insecure=port. My understanding is the former will match any incoming connection. The latter will match only those from the specified IP address. I could well be wrong as the documentation sucks.
edited once more to add:
I misunderstood the documentation. I now think you should be using insecure=port,invite.

Carriexchange do not use password based authentication. So if you are using "secret" then something is wrong without your trixbox configuration.

For those that have this issue.. I resolved it by installing the G729 codec issue resolved.

Anyone having issues with Carriex keep on dropping registration/unable to make calls?

I have tried to put in qualify=no and took out the registration string as advised by the tech support group but no luck.

Mine will work but every once in awhile it will drop registration/unable to make calls.

If i put qualify=no to the sip config for this trunk, then it would attempt to call but then it will fail after about 1 minute

Sorry for the delay but I have been out of town. I have a question for you sonvan. Are you using asterisk 1.2 or 1.4. I don not have a lot of experience with 1.4 yet, but I do know that several of my colleagues here that are using 1.4 have been having the same problem. My production system is running on 1.2 and with identical configurations I too am experiencing problems with 1.4.

The codec issue you resolved is always the first step in properly configuring your trunk for Carriex. I spent a great deal of time on the phone with Riza, their lead tech; he tells me that they truly ONLY support g729. At time you may get a call through with g711 and a particular sub provider but they do not support it as of the end of 2007.

I hope we can find out what is going on with Asterisk 1.4 that causes the problems we have been experiencing.

Have a great day!

hello
need help with carriex,
calls go thru but no audio,
the follwoing is my config:
allow=ulaw&g729
canreinvite=no
context=from-pstn
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=64.71.145.237
insecure=very
nat=no
restrictcid=no
type=peer

i am still not able to connect to Carrierx, you guys saying that we should have G729 installed, why is that? Carrierx support G711 and other codecs too.

I am able to pass calls to CarrierX ( that is, I could hear it ring, then when it is answered, I am unable to hear anything at all.

I am using Callcentric for my inbound trunk, then A2billing processes it and then selects the Carrier X trunk. I get the normal voice prompts for for PINS, and the number to enter, the destination number rings, but then afterwards I get no audio.

Just before the ringing, this is what I see in my CLI

SIP/Carriex-0913ec90 answered SIP/66.193.176.35-09140608

then no audio.

my firewall is opened for 10000-30000 for UDP for RTP.



I have tried this setp, using other trunks ( Callcentric) using only g729 codec and that works. So I know it is not a codec related issue.

Anything I am not doing correctly? Any help would be very much appreciated.


Here's the output of my CLI ( with Trunk ID replaced by YYYY)

<------------->
--- (12 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 64.71.145.237:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK2c6e66f1;rport
From: "14164450218" <sip:[email protected]>;tag=as4b079e17
To: <sip:[email protected]>;tag=373630313966303
Call-ID: [email protected]
Contact: <sip:[email protected]>
CSeq: 102 INVITE
Accept: application/sdp
Max-Forwards: 70
User-Agent: IXC SS
Content-Type: application/sdp
Content-Length: 208

v=0
o=- 1224175378 1224175378 IN IP4 64.71.145.237
s=SS SIP Session
c=IN IP4 64.71.145.237
t=0 0
m=audio 24228 RTP/AVP 18
c=IN IP4 64.71.145.237
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 18
Peer audio RTP is at port 64.71.145.237:24228
Found audio description format G729 for ID 18
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 64.71.145.237:24228
-- SIP/Carriex-0913ec90 is ringing
-- SIP/Carriex-0913ec90 is making progress passing it to SIP/66.193.176.35-09140608
pbx*CLI>
<--- SIP read from 64.71.145.237:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK2c6e66f1;rport
From: "14164450218" <sip:[email protected]>;tag=as4b079e17
To: <sip:[email protected]>;tag=373630313966303
Call-ID: [email protected]
Contact: <sip:[email protected]>
CSeq: 102 INVITE
Accept: application/sdp
Max-Forwards: 70
User-Agent: IXC SS
Content-Type: application/sdp
Content-Length: 208

v=0
o=- 1224175378 1224175378 IN IP4 64.71.145.237
s=SS SIP Session
c=IN IP4 64.71.145.237
t=0 0
m=audio 24228 RTP/AVP 18
c=IN IP4 64.71.145.237
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20

<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 18
Peer audio RTP is at port 64.71.145.237:24228
Found audio description format G729 for ID 18
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 64.71.145.237:24228
-- SIP/Carriex-0913ec90 is ringing
-- SIP/Carriex-0913ec90 is making progress passing it to SIP/66.193.176.35-09140608
pbx*CLI>
<--- SIP read from 64.71.145.237:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK2c6e66f1;rport
From: "14164450218" <sip:[email protected]>;tag=as4b079e17
To: <sip:[email protected]>;tag=373630313966303
Call-ID: [email protected]
Contact: <sip:[email protected]>
CSeq: 102 INVITE
Accept: application/sdp
Max-Forwards: 70
User-Agent: IXC SS
Content-Type: application/sdp
Content-Length: 264

v=0
o=- 1224175387 1224175387 IN IP4 64.71.145.237
s=SS SIP Session
c=IN IP4 64.71.145.237
t=0 0
m=audio 24228 RTP/AVP 18 101
c=IN IP4 64.71.145.237
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 64.71.145.237:24228
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.71.145.237:24228
list_route: hop: <sip:[email protected]>
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 64.71.145.237, port 5060
Transmitting ([email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK1d3cb9e1;rport
From: "14164450218" <sip:[email protected]>;tag=as4b079e17
To: <sip:[email protected]>;tag=373630313966303
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/Carriex-0913ec90 answered SIP/66.193.176.35-09140608

CarrieX = good luck IMO