vicidial.org

HI all,

Can anyone help me with direct SIP connection?
We had been having issues of voice breakage and call drops while in TPV so i contacted the TPV provider and they replied back saying we need to setup a Direct SIP connection. this is something new for me. Can anyone explain how to do it in step by step fashion? A bit of description on how a direct SIP works and what is the difference from regular calls? will really help me understand the situation...

This is my first post here so please ignore if i broke any rules... Thank you!

I would consider myself neither novice nor an expert in Dialer stuff...

Please help me out!! its urgent for me to solve this issue.!!!

Hi you need to request a direct connect or point to point connection from your sip provider! This is an interconnect direct into the pstn and often different providers use different terminology

Sip and the internet don't work well together so the internet needs bypassing

iainpg wrote:Hi you need to request a direct connect or point to point connection from your sip provider! This is an interconnect direct into the pstn and often different providers use different terminology

Sip and the internet don't work well together so the internet needs bypassing

hi can you clarify a bit more ??? We are using VICIbox...

this is what we have been asked to do by our TPV provider

You will need to set up a direct SIP connection with your dialer, VICI:

Hostname: voip-pos-e.ashergroup. com
Port: 5070

They must include the 10 digit TPV phone number in the SIP INVITEs.

Please be sure to use the hostname, rather than the IP address, as the
address may change.

If using asterisk (or equivalent), be sure that "qualify" is set to "no" (which
is probably the default)

what exactly should we request from our SIP provider. as so far none of them have been able to understand our requirements.

[carrier]
host=voip-pos-e.ashergroup. com
type=peer
disallow=all
allow=ulaw
allow=alaw
qualify=no
context=trunkinbound

boybawang wrote:[carrier]
host=voip-pos-e.ashergroup. com
type=peer
disallow=all
allow=ulaw
allow=alaw
qualify=no
context=trunkinbound

Hi thank you for the reply, currently testing the settings and will revert back if found any issue. so far going good...

Really thank you.

UGH we have this same problem!!! we get the same reply from Trusted TPV

kfadia wrote:

iainpg wrote:this is what we have been asked to do by our TPV provider

You will need to set up a direct SIP connection with your dialer, VICI:
Hostname: voip-pos-e.ashergroup. com
Port: 5070
They must include the 10 digit TPV phone number in the SIP INVITEs.
Please be sure to use the hostname, rather than the IP address, as the
address may change.
If using asterisk (or equivalent), be sure that "qualify" is set to "no" (which
is probably the default)
what exactly should we request from our SIP provider. as so far none of them have been able to understand our requirements.

same here.. our SIP provider has no idea what were talking about..
this post by Boybawang.. where would i put these settings.. Ty for you help... i am going nuts

boybawang wrote:[carrier]
host=voip-pos-e.ashergroup. com
type=peer
disallow=all
allow=ulaw
allow=alaw
qualify=no
context=trunkinbound

port=5070

very important

so your saying Add Port=5070 to the line.

[carrier]
host=voip-pos-e.ashergroup. com
type=peer
disallow=all
allow=ulaw
allow=alaw
qualify=no
context=trunkinbound
port=5070

and where do i do this, in the add carrier section

in the account entry field of the carrier add

on the account set up for Trusted TPV... what would be needed for theses 3 sections

Registration String:

Globals String:

Dialplan Entry:

if you have no user name and password, you are not expected to register, so registration string will be blank. this is normal for outbound only call connections.

globals and dialplan will be fairly standard but should be designed not to interfere with your existing dialplan or global string.

which brings me to the point: what are your existing globals string and dialplan settings for your existing carrier?

and:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

-----------

Remember: As far as your system is concerned, you are setting up a carrier. The fact that the carrier is a tpv does not change how your server views this generic outbound call. It only changes the IP to which it will send the call. It neither knows nor cares how the call will be answered or treated upon arrival. Your system considers this a standard outbound call. Treat it as such.

SOrry guys for late reply... Sollution provided by boybawang is working very well done very extensive testing.
so far i found port number is not required. i've tried this solution on multiple servers and working really good.

I am also using TrustedTPV and have the exact same requirements as the initial poster.

Hostname:
Port: 5070

They must include the 10 digit TPV phone number in the SIP INVITEs.

Please be sure to use the hostname, rather than the IP address, as the
address may change.

If using asterisk (or equivalent), be sure that "qualify=no" (which
is probably the asterisk default, but may be overridden), and
"dtmfmode=rfc2833". For audio encoding, we only provide g711u,
g711a and gsm. (g729 is not available.)


However, the tips in this thread are not working for me. My calls are still going out over the carrier and not directly connecting to their IP.

Any help would be greatly appreciated as we are losing customers left and right

TPV Calls are transfers. How are you transferring the calls to the TPV? (Button for button, as there are several methods available).

Your goal is to set up a Carrier (Admin->Carriers) and then cause the calls to the TPV to use that new carrier.

This is eventually a question of dialplan. Your new carrier (for the TPV) must have a dialplan entry that the calls to the TPV use because of a dial prefix, usually.