outbound proxy

New features you'd like see on SIP Sorcery
nederfox
Posts: 5
Joined: Sun Aug 26, 2007 11:13 am

outbound proxy

Post by nederfox » Sun Aug 26, 2007 11:30 am

Hi,
It would be lovely to have the "outbound proxy" feature. :roll:
That way we would be able to make calls with voicestick (i2telecom.com). They require to send <dialled_telephonenumber@i2telecom.com> to their outbound proxy 206.165.50.116.
Since last month, SIP requests sent to i2telecom.com are just dropped.

pablo
Posts: 54
Joined: Thu Jul 12, 2007 11:01 pm

Post by pablo » Sun Aug 26, 2007 3:47 pm

Something else comes to mind.

My IP device (Grandstream ATA or IP phone) has the following fields available :
SIP Server: (e.g., sip.mycompany.com, or IP address)
Outbound Proxy: (e.g., proxy.myprovider.com, or IP address)
SIP User ID: (the user part of an SIP address)
Authenticate ID: (can be same or different from SIP UserID)
It would be nice if SipSwitch could emulate all 4 fields, although I do realize that in most cases the fields are not all required.

Thanks

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Sun Aug 26, 2007 4:57 pm

Hi nederfox and pablo,

Both your requests have now been implemented and the SwitchCall command can take two additional parameters that allow the SIP From header and the destination for the request to be specified.

The new modifications are documented at the link below:

https://www.mysipswitch.com/helpinfo/di ... thelp.html

A quick example of a new command is:

exten => _X.,1,Switch(authuserid,pass,${dst}@sip.blueface.ie, "Joe Bloggs" <sip:sipuserid@blueface>, 194.213.29.100:5060)

Pablo with regards to the mappings on your phone:

SIP Server = the hostname portion of the third parameter, in the above example sip.blueface.ie,
OUtbound Proxy = the fifth parameter in the above example 194.213.29.100:5060,
SIP User ID = the user portiong of the SIP uri in the fourth parameter, in the above example sipuserid,
Authenticate ID = the first parameter, in the above example authuserid.

One thing worth noting is that I have yet to come across a SIP provider that operates with the SIP User and Auth IDs being different and in fact you will more than likely run into problems if you set them differently and call authentication is required.

Regards,

Aaron

pablo
Posts: 54
Joined: Thu Jul 12, 2007 11:01 pm

Post by pablo » Sun Aug 26, 2007 6:14 pm

Thanks, Aaron. That was fast !!
When I get a chance I'll go through the changes and digest it.
I'll let you know how it works out.

nederfox
Posts: 5
Joined: Sun Aug 26, 2007 11:13 am

Post by nederfox » Wed Aug 29, 2007 3:07 pm

Hi Aaron, that new implementation was lightning fast indeed!

Last week I tried to configure some providers within the Dialplan but that was not successful.

X-lite is registering ok with sipswitch
Also, zyxel P-2602HW-D1A and Sipura 3k are registering ok
Calling with other providers (voipbuster, fwd, messagenet, ...) works flawlessly but unfortunately, outgoing calls to mysipswitch will not go through.

On all calls X-lite reports Call failed: 404 NotFound. Even the 100 and 101 numbers are "not found". It might be some fault I made with the Dialplan config within the mysipswitch site. Could you have a quick view on this?

edit| code (replaced external ip with ***):

Code: Select all

SEND TIME: 18070843
SEND >> 213.200.94.182:5060
INVITE sip:101@sip.mysipswitch.com SIP/2.0
Via: SIP/2.0/UDP 62.51.43.***:13266;rport;branch=z9hG4bKB519918289EB41B6940EF8130E25489A
From: nederfox <sip:nederfox@sip.mysipswitch.com:7000>;tag=1321333541
To: <sip:101@sip.mysipswitch.com>
Contact: <sip:nederfox@62.51.43.***:13266>
Call-ID: 4253860D-1FEF-494E-81EC-05A42735315F@192.168.1.155
CSeq: 27443 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 222

v=0
o=nederfox 18070609 18070843 IN IP4 62.51.43.***
s=X-Lite
c=IN IP4 62.51.43.***
t=0 0
m=audio 13271 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE TIME: 18070890
RECEIVE << 213.200.94.182:5060
SIP/2.0 404 NotFound                        // Here's the issue  <<<< 
Via: SIP/2.0/UDP 62.51.43.***:13266;branch=z9hG4bKB519918289EB41B6940EF8130E25489A
To: <sip:101@sip.mysipswitch.com>
From: nederfox <sip:nederfox@sip.mysipswitch.com:7000>;tag=1321333541
Call-ID: 4253860D-1FEF-494E-81EC-05A42735315F@192.168.1.155
CSeq: 27443 INVITE
User-Agent: obelisk-sipproxy


SEND TIME: 18070906
SEND >> 213.200.94.182:5060
ACK sip:101@sip.mysipswitch.com SIP/2.0
Via: SIP/2.0/UDP 62.51.43.***:13266;rport;branch=z9hG4bKB519918289EB41B6940EF8130E25489A
From: nederfox <sip:nederfox@sip.mysipswitch.com:7000>;tag=1321333541
To: <sip:101@sip.mysipswitch.com>
Contact: <sip:nederfox@62.51.43.***:13266>
Call-ID: 4253860D-1FEF-494E-81EC-05A42735315F@192.168.1.155
CSeq: 27443 ACK
Max-Forwards: 70
Content-Length: 0

*** Log updates disabled ***

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Fri Aug 31, 2007 12:39 pm

Hi nederfox,

Could you post up the lines in your dialplan (minus sensitive bits) that you are using to match the 100 and 101 calls?

Regards,

Aaron

lothario
Posts: 6
Joined: Fri Aug 31, 2007 2:08 am

Post by lothario » Fri Aug 31, 2007 9:04 pm

I have the same problem...

My dial plan for example, when I want to call (outgoing call) to Brazil:

exten => _*2X.,1,Switch(myloginonvoipprovider,password,${EXTEN:2}@vono.net.br)

*2 is the prefix

vono.net.br is the sip server, but when I call *2+Number of Brazil

Have this problem, and I don´t understand:

Switching sip:*20@sip.mysipswitch.com:5060->sip:0@vono.net.br:5060 via vono.net.br.
21:57:22:999: Response Proxy Authentication Required from vono.net.br.
21:57:23:270: Response Address Incomplete from vono.net.br.


Am I have been something wrong??

nederfox
Posts: 5
Joined: Sun Aug 26, 2007 11:13 am

Post by nederfox » Fri Aug 31, 2007 9:07 pm

Initially I modified the dialplan. To investigate the 404 error, I tried with the default dialplan, which is also not succesful in solving the puzzle...
; Example extensions
exten = 100,1,Switch(anon,,303@sip.blueface.ie)
exten = 101,1,Switch(anon,,612@fwd.pulver.com)

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Fri Aug 31, 2007 10:53 pm

Hi lotahrio,

Address incomplete normally means the number you have sent to your SIP server is not recognised, i.e. the equivalent of a wrong number. The monitoring screen should show you a message something like:

"Switching *21234 to 1234@server"

Hth,

Aaron

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Fri Aug 31, 2007 11:13 pm

nederfox wrote:Initially I modified the dialplan. To investigate the 404 error, I tried with the default dialplan, which is also not succesful in solving the puzzle...
; Example extensions
exten = 100,1,Switch(anon,,303@sip.blueface.ie)
exten = 101,1,Switch(anon,,612@fwd.pulver.com)
Hi nederfox,

The above dialpan won't work as the 100 extension does not exist for Blueface and I'm pretty sure 101 does not exist for FreeWorldDialup either.

I suspect what you need is something along the lines of:

exten => _*1X.,1,Switch(username,password,${EXTEN:2}@youserver.com)

Hth,

Aaron

Post Reply