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309 posts

Ultimate Geek
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Topic # 29493 9-Jan-2009 09:56
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Hi,

I'm using trixbox and have 2 numbers with Xnet, inbound is fine however outbound is not.
I seem to get about 6-8 seconds of voice then it fails, it would appear that XNET are not receiving my ACK signal, however i'd like to confirm this with someone that has access to the core nodes if possible.

When the call is placed, I see the normal invites etc, then I get an answer and voice, however CLI of asterisk doesnt show the call as connected, then all of a sudden I get:
    -- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("SIP/1020-b7d09400", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack

My 5 italk numbers don't seem to have a problem.

Hopefully this will be of help?
<--- SIP read from 58.28.20.150:5060 --->
SIP/2.0 183 Session Progress
From: "94409877"<sip:94409877@as.wxcnz.net>;tag=as4941b0b1
To: <sip:0800103060@as.wxcnz.net>;tag=96141c3a-13c4-4966562d-e70ac63a-35a70817
Call-ID: 4faadf3b057906515f25a1c303b6166d@as.wxcnz.net
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Session: Media
Remote-Party-ID: <sip:0800103060@10.251.1.11;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
Via: SIP/2.0/UDP 113.21.224.43:5060;rport=5060;branch=z9hG4bK0884b1e6
Contact: <sip:0800103060@as.wxcnz.net:5060;maddr=58.28.20.150;transport=UDP>
Content-Type: application/sdp
Content-Length: 227

v=0
o=BroadWorks 21732196 1 IN IP4 58.28.20.150
s=-
c=IN IP4 58.28.20.150
t=0 0
m=audio 36450 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=fmtp:101 0-16
a=bsoft:1 image udptl t38

<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:36450
Found audio description format telephone-event for ID 101
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 58.28.20.150:36450
    -- SIP/KunzeaPL-09d640e8 is making progress passing it to SIP/099742910-b7d09400
trixbox*CLI> qui
<--- SIP read from 58.28.20.150:5060 --->
BYE sip:94409877@113.21.224.43 SIP/2.0
From: <sip:0800103060@as.wxcnz.net>;tag=96141c3a-13c4-4966562d-e70ac63a-35a70817
To: "94409877"<sip:94409877@as.wxcnz.net>;tag=as4941b0b1
Call-ID: 4faadf3b057906515f25a1c303b6166d@as.wxcnz.net
CSeq: 1 BYE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-1e1836-49665635-e70ae25d-1963f53f
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Sending to 58.28.20.150 : 5060 (no NAT)

<--- Transmitting (no NAT) to 58.28.20.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-1e1836-49665635-e70ae25d-1963f53f;received=58.28.20.150
From: <sip:0800103060@as.wxcnz.net>;tag=96141c3a-13c4-4966562d-e70ac63a-35a70817
To: "94409877"<sip:94409877@as.wxcnz.net>;tag=as4941b0b1
Call-ID: 4faadf3b057906515f25a1c303b6166d@as.wxcnz.net
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:94409877@113.21.224.43>
Content-Length: 0




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

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309 posts

Ultimate Geek
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  Reply # 188345 9-Jan-2009 10:21
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Sorry had to disable ALG support on the Juniper like i did last time, found an old post from myself :P

Wonder why Italk works fine but not Xnet?




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

199 posts

Master Geek
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  Reply # 188347 9-Jan-2009 10:42
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Because iTalk's platform is totally different - each device you connect on iTalk only communicates with one server on their side.  WorldxChange run Broadsoft which is a distributed platform that is made up of border controllers (what you're registering your device to), and application servers (the as.wxcnz.net address you see) which provide all of the functionality like voicemail, call forwarding etc.  In a setup like this, ALG features will modify the packets your device is sending to "support" NAT, and so when the packet reaches the border controller it is much different to what was expected when you registered.  Your registration will be going out with your internal IP address, which WxC is now expecting to talk to, but when you send an INVITE through an ALG the IP address on it is probably being changed to your external IP, and so when Broadsoft gets that packet it discards it because it has no idea who it is.

SIP ALG's are only really designed to be used in enterprise VoIP networks, and as you'll be aware Juniper/Cisco routers are intended for use there, hence having such a feature.  In an enterprise network your branch office is usually linked back to some sort of VoIP router in that area's main office, like an Avaya IP Office (if you're in Juniper land) or Call Manager (if you're in Cisco land).  They aren't really there for use in service provider scenarios.


If you want to understand the issue more, login to your VFX portal and look under Registrations, then compare it to the output you pasted below.  You'll probably find that the header in there doesn't match what you're sending below because the ALG was modifying it.

 
 
 
 




309 posts

Ultimate Geek
+1 received by user: 5

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  Reply # 188348 9-Jan-2009 10:49
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I've actually been trying to login to the portal, I sent an email yesterday for a password reset but heard nothing back yet...

Secondly the registration is actually coming from a colo box with a live IP, so the NAT must be on your side perhaps?

I'm having a totally different issue now, when making outbound calls, I never receive ring tone, 90% of calls when answered I get voice, but others I dont, any idea what that could be caused by?

thanks




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone



309 posts

Ultimate Geek
+1 received by user: 5

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  Reply # 188352 9-Jan-2009 10:55
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In addition, I have the following ports enabled:

Custom1: udp 16384-53999
Custom2: 5060
junos-rtsp - tcp 554
junos-sip




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone



309 posts

Ultimate Geek
+1 received by user: 5

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  Reply # 188355 9-Jan-2009 11:02
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I just added another rule to allow the Xnet-VfX to get to the trixbox for all services, audio seems to be ok, but i'll keep testing, still no ring though, any ideas?




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

199 posts

Master Geek
+1 received by user: 7

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  Reply # 188359 9-Jan-2009 11:11
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It's not my side, I don't work for them :)

That said, the helpdesk can reset your password over the phone if you call 0800 14 96 38 and talk to technical support.  It only takes a few seconds.
If you're not operating NAT on your side and you run an ALG, the IP will be getting changed to the IP of the Netscreen, not the IP of your Asterisk server.  If you have that turned off now (you wouldn't need it if you don't run NAT anyway), any issue is going to be something to do with packets not getting through, which generally points to a firewall somewhere.

If you aren't getting ringing on your phones when dialing out that probably means you aren't getting progress/ringing messaging back on your end.  I don't deal with Juniper kit but the rules you would need on IOS are:

permit udp host 58.28.20.150 eq 5060 anypermit udp host 58.28.20.150 range 16384 53999 any
E.g., allow udp 5060+16384-53999 from 58.28.20.150 to ANY on your side

205 posts

Master Geek
+1 received by user: 1

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WorldxChange

  Reply # 188898 12-Jan-2009 08:32
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Hi icepick,

What you will need to do is trace the SIP packets on your asterisk box back to your phone, see if you are getting the ring message i.e. 180/183.

You can do this by logging into the command line and typing the following:

ngrep -d eth0 port 5060 and host 192.168.x.x

change the eth0 for the ethernet port that you are using and the 192.168 address for the local IP of the handset.

This will give you the call flow when a call is being sent to the phone.  If the messages are getting to the phone you will need to see if the message is a 180 or a 183, a 180 will tell the device/phone to generate the ringing where as the 183 will tell the device to open up the rtp stream to play back the ringing from the asterisk box.

So based on that fact, either the phone may not be generating the ringtone or the audio path is not being opened up to/from the asterisk box for ringing.

Also try using another phone, try perhaps downloading xlite and registering that, see if you can replicate the fault.

Cheers

Josh




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