outbound proxy
outbound proxy
Hi,
It would be lovely to have the "outbound proxy" feature.
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.
It would be lovely to have the "outbound proxy" feature.
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.
Something else comes to mind.
My IP device (Grandstream ATA or IP phone) has the following fields available :
Thanks
My IP device (Grandstream ATA or IP phone) has the following fields available :
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.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)
Thanks
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,${dst}@sip.blueface.ie, "Joe Bloggs" <sip:sipuserid@blueface>, 194.213.29.100:5060)
Pablo with regards to the mappings on your phone:
SIP Server = the hostname portion of the third parameter, in the above example sip.blueface.ie,
OUtbound Proxy = the fifth parameter in the above example 194.213.29.100:5060,
SIP User ID = the user portiong of the SIP uri in the fourth parameter, in the above example sipuserid,
Authenticate ID = the first parameter, in the above example authuserid.
One thing worth noting is that I have yet to come across a SIP provider that operates with the SIP User and Auth IDs being different and in fact you will more than likely run into problems if you set them differently and call authentication is required.
Regards,
Aaron
Both your requests have now been implemented and the SwitchCall command can take two additional parameters that allow the SIP From header and the destination for the request to be specified.
The new modifications are documented at the link below:
https://www.mysipswitch.com/helpinfo/di ... thelp.html
A quick example of a new command is:
exten => _X.,1,Switch(authuserid,pass,${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
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 ***):
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 ***
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,${EXTEN:2}@vono.net.br)
*2 is the prefix
vono.net.br is the sip server, but when I call *2+Number of Brazil
Have this problem, and I don´t understand:
Switching sip:*20@sip.mysipswitch.com:5060->sip:0@vono.net.br:5060 via vono.net.br.
21:57:22:999: Response Proxy Authentication Required from vono.net.br.
21:57:23:270: Response Address Incomplete from vono.net.br.
Am I have been something wrong??
My dial plan for example, when I want to call (outgoing call) to Brazil:
exten => _*2X.,1,Switch(myloginonvoipprovider,password,${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??
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)
; Example extensions
exten = 100,1,Switch(anon,,303@sip.blueface.ie)
exten = 101,1,Switch(anon,,612@fwd.pulver.com)
Hi nederfox,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)
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