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Guideline for SIP Trunk Setup

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Guideline

Guideline

Guideline

Guideline

for

for

for

for

SIP

SIP

SIP

SIP Trunk

Trunk

Trunk

Trunk Setup

Setup

Setup

Setup

with

(2)

Table

Table

Table

Table of

of

of

of contents

contents

contents

contents

Sample sip.conf (it applies to asterisk 1.4.x)...3

Sample elastix setup... 3

Ports required... 4

Caller ID... 4

FAQ... 5

After i dial out, the destination phone rings, but we could not hear each other... 5

How to dial to HK?... 5

How to make international call?...5

Sometimes, I could not receive incoming calls...6

I do not get a dial tone from zonetel...6

I can dial to HK, but not other destinations...6

The voice quality is not good... 6

I can dial out directly from phone, but call forward fails...7

(3)

Sample

Sample

Sample

Sample sip.conf

sip.conf

sip.conf

sip.conf (it

(it

(it

(it applies

applies

applies

applies to

to

to

to asterisk

asterisk

asterisk

asterisk 1.4.x)

1.4.x)

1.4.x)

1.4.x)

[general] Defaultexpiry=600 Register => 5804xxxx:secret@zonetel:5060/5804xxxx~600 [zonetel] Username=5804xxxx Fromuser=5804xxxx Secret=<we provided> Host=1511.zonetel.com Type=friend Insecure=very Qualify=no Nat=yes Canreinvite=no Disallow=all Allow=alaw Allow=ulaw Permit=202.130.146.96/255.255.255.224

Sample

Sample

Sample

Sample elastix

elastix

elastix

elastix setup

setup

setup

setup

1.Basic>Trunks>Add Trunk Outbound callerid = 5804xxxx Dial Rules = . Peer details= host=1511.zonetel.com type=peer permit=202.130.146.96/255.255.255.224 Defaultexpiry=600

register string=5804xxxx:your sip

(4)

2.Basic>Outbound Routes>Add Routes ;; if you dial 9 to call out via sip, etc. Dial Patterns=

9|.

Trunk Sequence=<the trunk created in 1>

3.Inbound Call Control>Inbound Routes>Add Incoming Route DID number=5804xxxx

Extensions=<the extension to answer inbound call from 5804xxxx>

Ports

Ports

Ports

Ports required

required

required

required

Please check that your firewall allows below outgoing traffic to our network. � 5060 tcp&udp for sip signaling

� 40000-44000 udp to 202.130.146.96/255.255.255.224 for calling HK � All udp to 202.130.146.96/255.255.255.224 for calling IDD

Caller

Caller

Caller

Caller ID

ID

ID

ID

(5)

FAQ

FAQ

FAQ

FAQ

After

After

After

After iiii dial

dial

dial

dial out,

out,

out,

out, the

the

the

the destination

destination

destination

destination phone

phone

phone

phone rings,

rings,

rings,

rings, but

but

but

but we

we

we

we

could

could

could

could not

not

not

not hear

hear

hear

hear each

each

each

each other.

other.

other.

other.

� Check that you don't have any firewall rules blocking traffic from/to our ip 202.130.146.96/255.255.255.224

� Please check that your firewall allows udp traffic to your SIP

server/endpoint. If you are using Asterisk, you can check its RTP port range in /etc/asterisk/rtp.conf.

� One way audio could be due to problems with NAT handling. Please check the nat and canreinvite parameters in your sip configuration. � Please check that you are using g711 (alaw or ulaw).

� By default, we support g711. For g729 support, please make prior arrangement with our service team.

How

How

How

How to

to

to

to dial

dial

dial

dial to

to

to

to HK?

HK?

HK?

HK?

� Simply use 8-digits as the HK destination number (the '852' country prefix is not necessary). The callee will see your 5804xxxx as caller id.

� Outgoing calls to HK is free of charge.

� Please make sure your are using 5804xxxx as the callerid

How

How

How

How to

to

to

to make

make

make

make international

international

international

international call?

call?

call?

call?

� In our platform, we treat all non-HK calls as international call.

� Use dial string 1511+country+area+destination to make international calls. � Call charges apply. Please contact us for rate plan.

� Please make sure you are using 5804xxxx as the callerid

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Sometimes,

Sometimes,

Sometimes,

Sometimes, IIII could

could

could

could not

not

not

not receive

receive

receive

receive incoming

incoming

incoming

incoming calls.

calls.

calls.

calls.

� It could be due to sip registration being expired too early. We recommend a sip registration TTL value of at least 600.

� Please check that 5804xxxx is being used by one sip end point only. (This applies when your sip end point directly registers to our server rather than your own PBX)

IIII do

do

do

do not

not

not

not get

get

get

get a

a dial

a

a

dial

dial

dial tone

tone

tone

tone from

from

from

from zonetel

zonetel

zonetel

zonetel

Please check that your firewall permits traffic to ports 40000 - 44000 of 202.130.146.96/27.

IIII can

can

can

can dial

dial

dial

dial to

to

to

to HK,

HK,

HK, but

HK,

but

but

but not

not

not

not other

other

other

other destinations

destinations

destinations

destinations

� Please allow all UDP ports from our subnet 202.130.146.96/27. The 40000-44000 is for HK calling. We will use any other UDP ports for international calling.

� By default, 5804xxxx is IDD-disabled. Please contact our service team to open the IDD service and inquire the IDD rates.

� We highly recommend IDD PIN. Please contact our service team for the procedures.

The

The

The

The voice

voice

voice

voice quality

quality

quality

quality is

is

is

is not

not

not

not good

good

good

good

(7)

be used

IIII can

can

can

can dial

dial

dial

dial out

out

out

out directly

directly

directly

directly from

from

from phone,

from

phone,

phone,

phone, but

but

but

but call

call

call

call forward

forward

forward

forward fails

fails

fails

fails

When your PBX forwards call via our SIP trunk, please ensure that it is sending 5804xxxx as the callerid. Outside caller resulted from a forwarded call will not be recognized by us. If you are using Elastix, please choose trunk CID options 'Block Foreign CID' or 'Force Trunk CID'.

Still

Still

Still

Still have

have

have

have problem?

problem?

problem?

problem?

Please send us the following information for troubleshooting. � Sip.conf

� Sip show peer <peer name>

References

Related documents

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