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Cannot register SIP trunk with talkinip

I am trying to create a SIP trunk to talkinip, however it doesnt seem to be working and i cannot see any error messages.

I have attached my sip.conf

debug = 9
sip debug is on

show sip registry - shows nothing

when i dial the trunk i get this on the console.

^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 0: SIP/2.0 403 Forbidden (21)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 1: To: <sip:CELL Number@64.154.41.200>;tag=3425206729-283278 (57)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 2: From: "User" <sip:Username@Externip>;tag=as2ef64c04 (63)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 3: Call-ID: 4dd1fa752c6d8d4331eca4e973e0e6bc@externip (55)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 4: CSeq: 102 INVITE (16)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 5: Via: SIP/2.0/UDP externip:10165;rport;branch=z9hG4bK4679db3f (65)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 6: Content-Length: 0 (17)
^@[Jul 16 09:18:50] DEBUG[4445]: chan_sip.c:4776 parse_request: ^@Header 7: Contact: <sip:CellNumber@64.154.41.200> (40)

Thanks in advance

Ash
[general]
context=Default
externip= My IP
externrefresh=15
localnet=10.0.0.0/255.0.0.0
canreinvite=no
bindport=5060
bindaddr=10.4.10.2 ; Asterisk IP
disallow=all
allow=ulaw
allow=gsm
language=en
trustrpid=yes
sendrpid=yes
pickupgroup=1
callgroup=1
registerattempts=0 ; Just to make sure the rgister string keeps trying
 
register=username:secret@64.154.41.200/username ; from talkinip website
 
[authentication]
 
[talkinip-trunk]
username=7645030753
type=peer
secret=my secret
insecure=very
nat=no ; have tried with yes also
host=64.154.41.200
fromuser=my username
dtmfmode=rfc2833
canreinvite=no
allow=ulaw
context=default

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Member_2_1968385
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You should first investigate why Asterisk is failing to register with talkinip - if the registration is failing then an outbound call has little chance of working. Asterisk will keep attempting to register, but you can also force it to retry by doing "sip reload". Instead of using the ordinary Asterisk debug, I recommend you use the "sip debug" or "sip set debug" command at the CLI to make it display the SIP packets. Registration should show a series of packets exchanged, roughly along these lines (my examples are highly edited):
From Asterisk: REGISTER sip:domain SIP/2.0
To Asterisk:  SIP/2.0 100 Trying
To Asterisk:  SIP/2.0 401 Unauthorised
                     ......
                     WWW-Authenticate: Digest realm="domain", nonce="some_long_string_of_chars"
                     ......
From Asterisk: REGISTER sip:domain SIP/2.0
                        ......
                        Authorization: Digest username="user", realm="domain", algorithm=MD5, uri="sip:domain", nonce="long_string", response="another_long_string", opaque=""
                        ......
To Asterisk:  SIP/2.0 100 Trying
To Asterisk:  SIP/2.0 200 OK

The above is a typical Challenge authentication sequence, a process that most SIP Registrars will follow more or less.

I wonder if the problem is simply that you are using the IP address 64.154.41.200 instead of the host.domain name sip.talkinip.net. It is possible that it needs to see "sip.talkinip.net" as part of the registration/authentication and an IP address is not regarded as being equivalent. Also check for typos in your username and password - that was the problem in a recent question here.
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ASKER

sorry i was using sip set debug too but still cannot see the challenge authentication. I will check again.

The support guy asked me to try with the ip as the host name wasnt working either.

When i do a SIP Reload it still does not look like asterisk is even trying to register with talkinip, it goes straight into registering the handsets. I have attached the log.
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4564 find_call:  = Found Their Call ID: 285d209557d763cd5ec279636136b915@10.4.10.2 Their Tag  Our tag: as7dc02fcc
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:2162 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2104
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:2172 __sip_ack:  Stopping retransmission on '285d209557d763cd5ec279636136b915@10.4.10.2' of Request 102: Match Not Found
 
SIP*CLI> 
Really destroying SIP dialog '285d209557d763cd5ec279636136b915@10.4.10.2' Method: NOTIFY
 
<--- SIP read from 10.4.50.248:5060 --->
SIP/2.0 200 OK
 
To: <sip:3701@10.4.50.248:5060>;tag=46a964ba86f78667i0
 
From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as3317855e
 
Call-ID: 2e9c77c83cd1faff27656ddd4c797867@10.4.10.2
 
CSeq: 102 NOTIFY
 
Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK6a7831ea
 
Server: Linksys/SPA942-5.2.8
 
Content-Length: 0
 
 
 
 
<------------->
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 0: SIP/2.0 200 OK (14)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 1: To: <sip:3701@10.4.50.248:5060>;tag=46a964ba86f78667i0 (54)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 2: From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as3317855e (56)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 3: Call-ID: 2e9c77c83cd1faff27656ddd4c797867@10.4.10.2 (51)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 4: CSeq: 102 NOTIFY (16)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 5: Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK6a7831ea (54)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 6: Server: Linksys/SPA942-5.2.8 (28)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 7: Content-Length: 0 (17)
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 8:  (0)
 --- (8 headers 0 lines) ---
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:4564 find_call:  = Found Their Call ID: 2e9c77c83cd1faff27656ddd4c797867@10.4.10.2 Their Tag  Our tag: as3317855e
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:2162 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2108
 [Jul 17 02:45:34] DEBUG[4445]: chan_sip.c:2172 __sip_ack:  Stopping retransmission on '2e9c77c83cd1faff27656ddd4c797867@10.4.10.2' of Request 102: Match Not Found
 Really destroying SIP dialog '2e9c77c83cd1faff27656ddd4c797867@10.4.10.2' Method: NOTIFY
 
SIP*CLI> 
[Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4511 sip_alloc:  Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP)
 [Jul 17 02:45:35] DEBUG[4445]: acl.c:215 ast_apply_ha:  ##### Testing 10.4.50.249 with 10.4.0.0
 Scheduling destruction of SIP dialog '24a38484286a8e3470156a957c0677a8@10.4.10.2' in 32000 ms (Method: NOTIFY)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 0: NOTIFY sip:3001@10.4.50.249:5060 SIP/2.0 (40)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 1: Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK5846be95;rport (60)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 2: From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as2291cac8 (56)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 3: To: <sip:3001@10.4.50.249:5060> (31)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 4: Contact: <sip:asterisk@10.4.10.2> (33)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 5: Call-ID: 24a38484286a8e3470156a957c0677a8@10.4.10.2 (51)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 6: CSeq: 102 NOTIFY (16)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 7: User-Agent: Asterisk PBX (24)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 8: Max-Forwards: 70 (16)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 9: Event: message-summary (22)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 10: Content-Type: application/simple-message-summary (48)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 11: Content-Length: 89 (18)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 12:  (0)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4808 parse_request:  Line: Messages-Waiting: no (20)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4808 parse_request:  Line: Message-Account: sip:asterisk@10.4.10.2 (39)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4808 parse_request:  Line: Voice-Message: 0/0 (0/0) (24)
 Reliably Transmitting (NAT) to 10.4.50.249:5060:
NOTIFY sip:3001@10.4.50.249:5060 SIP/2.0
 
Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK5846be95;rport
 
From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as2291cac8
 
To: <sip:3001@10.4.50.249:5060>
 
Contact: <sip:asterisk@10.4.10.2>
 
Call-ID: 24a38484286a8e3470156a957c0677a8@10.4.10.2
 
CSeq: 102 NOTIFY
 
User-Agent: Asterisk PBX
 
Max-Forwards: 70
 
Event: message-summary
 
Content-Type: application/simple-message-summary
 
Content-Length: 89
 
 
 
Messages-Waiting: no
 
Message-Account: sip:asterisk@10.4.10.2
 
Voice-Message: 0/0 (0/0)
 
 
---
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:2043 __sip_reliable_xmit:  *** SIP TIMER: Initializing retransmit timer on packet: Id  #2110
 
SIP*CLI> 
 
<--- SIP read from 10.4.50.249:5060 --->
SIP/2.0 200 OK
 
To: <sip:3001@10.4.50.249:5060>;tag=725e515faa409c2i0
 
From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as2291cac8
 
Call-ID: 24a38484286a8e3470156a957c0677a8@10.4.10.2
 
CSeq: 102 NOTIFY
 
Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK5846be95
 
Server: Linksys/SPA942-5.2.8
 
Content-Length: 0
 
 
 
 
<------------->
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 0: SIP/2.0 200 OK (14)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 1: To: <sip:3001@10.4.50.249:5060>;tag=725e515faa409c2i0 (53)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 2: From: "asterisk" <sip:asterisk@10.4.10.2>;tag=as2291cac8 (56)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 3: Call-ID: 24a38484286a8e3470156a957c0677a8@10.4.10.2 (51)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 4: CSeq: 102 NOTIFY (16)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 5: Via: SIP/2.0/UDP 10.4.10.2:5060;branch=z9hG4bK5846be95 (54)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 6: Server: Linksys/SPA942-5.2.8 (28)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 7: Content-Length: 0 (17)
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4776 parse_request:  Header 8:  (0)
 --- (8 headers 0 lines) ---
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:4564 find_call:  = Found Their Call ID: 24a38484286a8e3470156a957c0677a8@10.4.10.2 Their Tag  Our tag: as2291cac8
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:2162 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2110
 [Jul 17 02:45:35] DEBUG[4445]: chan_sip.c:2172 __sip_ack:  Stopping retransmission on '24a38484286a8e3470156a957c0677a8@10.4.10.2' of Request 102: Match Not Found
 Really destroying SIP dialog '24a38484286a8e3470156a957c0677a8@10.4.10.2' Method: NOTIFY

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Perhaps you are not capturing the full exchange of SIP messages - there can be a lot of them especially just after issuing a "sip reload" instruction. The log you have posted above only shows some NOTIFY requests to tell the handsets whether to switch on the message waiting lamp.

To capture SIP packets in a more controlled way, you could consider installing a utility like sipgrep:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/utils/sipgrep

My preferred solution (other than sipgrep) is to capture the Asterisk SIP packet output in the Asterisk log file (/var/log/asterisk/messages) by setting these options in the logger.conf file:
[general]
dateformat=%F %T
[logfiles]
messages => error,verbose
Ok i have set those options and still i am only getting the notigy messages.

i have done logger reload and a full reload.
Can you force one of your handsets to re-register itself with Asterisk (e.g. by rebooting the phone) - just to make sure you are capturing REGISTER messages in the log file.

The sip.conf file that you posted above is very brief. I wonder if you have edited it down to the point where you have removed something vital. For example, there is usually at least one "domain=..." statement in the sip.conf [general] section. I think this is an unlikely explanation, but I'm really quite puzzled as to what could prevent Asterisk from trying to register itself.
ok here is the messages file after a full reload
messages.txt
ahd i could see register messages in that file for the individual handsets so the log file must be picking up on them
hmm. I can't see a reason for the registration not to happen. As you say, the log file has captured registration requests from the handsets with no problem. What is the actual output on the screen for the two CLI commands:
sip show peers
sip show registry
ok top half is "Sip show Peers"  the 81.*.*.* are the voxbone incoming peers

bottom half "SIP show registry"
Name/username              Host            Dyn Nat ACL Port     Status
3702/3702                  (Unspecified)    D   N      0        Unmonitored
3302/3302                  10.4.50.243      D   N      5060     Unmonitored
3802/3802                  10.4.50.245      D   N      5060     Unmonitored
3801/3801                  10.4.50.247      D   N      5060     Unmonitored
3201/3201                  10.4.50.246      D   N      5060     Unmonitored
3301/3301                  10.4.50.244      D   N      5060     Unmonitored
3701/3701                  10.4.50.248      D   N      5060     Unmonitored
3001/3001                  10.4.50.249      D   N      5060     Unmonitored
talkinip-trunk/7645030753  64.154.41.200               5060     Unmonitored
81.201.84.45               81.201.84.45         N      5060     Unmonitored
81.201.84.42               81.201.84.42         N      5060     Unmonitored
81.201.84.41               81.201.84.41         N      5060     Unmonitored
81.201.84.40               81.201.84.40         N      5060     Unmonitored
81.201.84.39               81.201.84.39         N      5060     Unmonitored
81.201.84.38               81.201.84.38         N      5060     Unmonitored
81.201.84.37               81.201.84.37         N      5060     Unmonitored
81.201.84.36               81.201.84.36         N      5060     Unmonitored
81.201.84.35               81.201.84.35         N      5060     Unmonitored
81.201.84.34               81.201.84.34         N      5060     Unmonitored
81.201.84.33               81.201.84.33         N      5060     Unmonitored
81.201.84.32               81.201.84.32         N      5060     Unmonitored
81.201.84.31               81.201.84.31         N      5060     Unmonitored
81.201.84.30               81.201.84.30         N      5060     Unmonitored
81.201.84.29               81.201.84.29         N      5060     Unmonitored
81.201.84.28               81.201.84.28         N      5060     Unmonitored
81.201.84.27               81.201.84.27         N      5060     Unmonitored
81.201.84.26               81.201.84.26         N      5060     Unmonitored
81.201.84.25               81.201.84.25         N      5060     Unmonitored
81.201.84.24               81.201.84.24         N      5060     Unmonitored
81.201.84.23               81.201.84.23         N      5060     Unmonitored
81.201.84.22               81.201.84.22         N      5060     Unmonitored
81.201.84.21               81.201.84.21         N      5060     Unmonitored
81.201.84.20               81.201.84.20         N      5060     Unmonitored
81.201.82.33               81.201.82.33         N      5060     Unmonitored
81.201.82.32               81.201.82.32         N      5060     Unmonitored
81.201.82.31               81.201.82.31         N      5060     Unmonitored
81.201.82.30               81.201.82.30         N      5060     Unmonitored
81.201.82.29               81.201.82.29         N      5060     Unmonitored
81.201.82.28               81.201.82.28         N      5060     Unmonitored
81.201.82.27               81.201.82.27         N      5060     Unmonitored
81.201.82.26               81.201.82.26         N      5060     Unmonitored
81.201.82.25               81.201.82.25         N      5060     Unmonitored
81.201.82.24               81.201.82.24         N      5060     Unmonitored
81.201.82.23               81.201.82.24         N      5060     Unmonitored
81.201.82.22               81.201.82.22         N      5060     Unmonitored
81.201.82.21               81.201.82.21         N      5060     Unmonitored
81.201.82.20               81.201.82.20         N      5060     Unmonitored
 
SIP*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time

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I have also checked with talkinip and they dont even see me trying to register either. But then that corresponds to what im looking at.

there must be something i have missed but i just cant see it.
There must be a problem in your sip.conf file. I just added a phony "register" entry into my sip.conf file, below the existing "register" that was already there and working. When I do "sip reload" it immediately gives me an error message saying that it cannot resolve the host name I used (because it is just some rubbish I made up). Also, when I do "sip show registry" it shows me the phony entry, but with the state as Unregistered, alongside the working entry.

You are not even seeing an entry in the list for talkinip when you use "sip show registry". Therefore it cannot be reading that line properly from your sip.conf file. Do you have a backup of sip.conf that you could use for testing - perhaps one of the original sample files from the asterisk install?
I have the original files.  so i will make a backup of this one and copy an original over to see what happens.
funnily enough if i put the sample config back in, the register string does show up in the debug info. how very weird.
not to give you something even more weird,  if i remove everything from that sip.conf apart from Generel and the register statement. it actually registers!!!!
ok i moved the #include voxbone.conf to below the register line

now everything works. how very weird.

thanks again for your help feptias you re awakened my brain again.

ash
I suppose you can't get consistent, fully tested open source products - there are always some weird bits in Asterisk. I have resorted to modifying the C source code to fix problems before now.

On the other hand, you'll probably be extremely lucky to get consistent, fully tested commercial software and it will have cost you a fortune too.

By the way, fixing the registration was only step 1. Don't you now have to check that you can make and receive calls through talkinip?
yeah thats why im leaving the post open :P the guys that use this phone server are in the states, so arent really awake yet
ok it works !!! they just cant hear the phone ringing but it does go through. i should be able to sort that out.
oh wait. no audio at all, both ends that is.
No audio is just a variation on the standard NAT traversal problem. Usually you get one-way audio.

What is the complete setup, end-to-end? Are they also using an Asterisk PBX behind NAT with a separate registration to talkinip? What handset are you using to make the test calls? Is it a locally registered IP phone? Do you have any POTS phones connected to your Asterisk? If so, try calling from one of them instead.
ok setup is .

Linksys SPA942 - asterisk PBX - Watchguard X550e fw 10.2 - dsl line - Talkinip

this is the same firewall as i have here in the uk.

The new york PBX ( i.e. the one having the probs) has two pots lines connected to a FXO card for backup, no fxs ports so i cant test pots phones.

incoming SIP calls work fine.
My box here is connected to theirs via iax2 which also works fine.

so the only problem i have is with talkinip outgoing SIP calls.
It is quite likely to be one of the following:
1. Re-invite telling the Linksys to talk directly to Talkinip, not via Asterisk. This will fail because your firewall will only be set to allow Asterisk to Talkinip
2. The firewall needs some ports opening to allow RTP media through.

The solution to 1: In the dialplan, find the relevant line where Dial is making the call to talkinip and add one of the following options to the Dial command - t, T", "h", "H", "w", "W" or "L". Any of these will force Asterisk to stay in the media path.

The solution to 2: Identify the RTP ports that Asterisk is using (Your earlier log file showed this as "RTP Allocating from port range 10000 -> 20000"). Then set the firewall for one-to-one NAT between externip and local Asterisk IP with UDP port forwarding allowed for that entire range of ports. If you want to have fewer ports open, you can reduce the range in the file rtp.conf. I would guess that you already have port forwarding on 5060.

Another thing to try, if the above solutuion to (2) is difficult to do or doesn't fix it, is to remove the line for externip from sip.conf and see if it helps. Sometimes, if the VoIP service provider has a "far-end NAT traversal solution" in place, then you may be fooling their SIP proxy server into thinking your Asterisk is not behind NAT because you told Asterisk to substitute its internal IP address with the externip value. Take that away and the far-end NAT traversal might kick in and solve the problem for you.
Another possibility is if Talkinip have different servers acting as media proxies to those used for SIP. It is quite common for a VoIP service provider to have several IP addresses and you may need to set your firewall to allow RTP access to/from all of them.
ok i added the -t to the dial string and i now have one way audio, i.e. caller can hear callee.

I am now going to put the externip back in
ok after adding externip back in, i get no audio again. but their internet is going up and down like a yoyo at the moment. so im sure thats not helping.
canreinvite=no

the firewall is forwarding udp ports 10000-20000 to the asterisk server ( from any for the time being)

i have tried with and without externip, and this still is not working.

But i am going to wait for their net to be more stable before i go any further.
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ok after lots of faffing with their internet connection.

the 1 to 1 Nat was not configured correctly on the watchguard, all is working.

thanks again feptias.