Getting destination number
Hello, I've just setup a sip trunk with a new provider, however there are some issues receiving calls. The INVITE looks like this: INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> ... The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-) Then Freeswitch does this of cause: [INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong? Best regards, Allan
Check out the info dialplan application… You’re provider is doing something unusual here as the invite URI is what you use to route the call, the to: uri is supposed to be the original dialed digits From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Allan Kristensen Sent: Wednesday, September 7, 2016 10:39 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Getting destination number Hello, I've just setup a sip trunk with a new provider, however there are some issues receiving calls. The INVITE looks like this: INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 <http://sip:NCdWXsdSsaF2B@192.168.0.190:52284> SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> ... The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-) Then Freeswitch does this of cause: [INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong? Best regards, Allan
https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Fil... Specifically the "extension" parameter. On 7 September 2016 at 16:43, Ken Rice <krice@freeswitch.org> wrote:
Check out the info dialplan application…
You’re provider is doing something unusual here as the invite URI is what you use to route the call, the to: uri is supposed to be the original dialed digits
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Allan Kristensen *Sent:* Wednesday, September 7, 2016 10:39 AM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* [Freeswitch-users] Getting destination number
Hello,
I've just setup a sip trunk with a new provider, however there are some issues receiving calls.
The INVITE looks like this:
INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 SIP/2.0
....
From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203
To: <sip:+YYYYYYYYYY@x.x.x.x>
...
The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-)
Then Freeswitch does this of cause:
[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound
That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ?
I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy...
Or is it my provider that is doing it wrong?
Best regards,
Allan
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I'd ask provider to make a change and send destination number in SIP URI like most others do. Especially if later you get different numbers from a different provider, you might benefit from having it all standard from day 1. On 16-09-07 11:38 AM, Allan Kristensen wrote:
Hello,
I've just setup a sip trunk with a new provider, however there are some issues receiving calls.
The INVITE looks like this:
INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 <http://sip:NCdWXsdSsaF2B@192.168.0.190:52284> SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> ...
The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-)
Then Freeswitch does this of cause:
[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound
That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong?
Best regards, Allan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It IS a trunk with multiple DID's and it IS registered with the provider. Setting the "extension" parameter however does not change anything. It's like the invite request is not "matched" with the gateway config and the option is therefore not applied. It's a pretty simple config: <include> <gateway name="dev_tx1_13"> <param name="proxy" value="x.x.x.x"/> <param name="ping" value="10"/> <param name="register" value="true"/> <param name="extension" value="auto_to_user"/> <param name="username" value="NCdWXsdSsaF2B"/> <param name="password" value="<censored>"/> </gateway> </include> I will contact the provider if the way they are doing it is non standard... /Allan On Wed, Sep 7, 2016 at 5:56 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
I'd ask provider to make a change and send destination number in SIP URI like most others do.
Especially if later you get different numbers from a different provider, you might benefit from having it all standard from day 1.
On 16-09-07 11:38 AM, Allan Kristensen wrote:
Hello,
I've just setup a sip trunk with a new provider, however there are some issues receiving calls.
The INVITE looks like this:
INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x> <sip:+XXXXXXXXXX@x.x.x.x>;tag= Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> <sip:+YYYYYYYYYY@x.x.x.x> ...
The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-)
Then Freeswitch does this of cause:
[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound
That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong?
Best regards, Allan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If this is to a registered trunk, and they include the tags we use in the registered contact, we will properly detect this, and use the to user instead of the request user. This allows you to register once for multiple DID’s. Otherwise what the provider is doing here is weird and pretty non standard, and you would need to handle it manually in the dial plan looking at the sip_to_user var. Mike
On Sep 7, 2016, at 11:38 AM, Allan Kristensen <ak@hejdu.dk> wrote:
Hello,
I've just setup a sip trunk with a new provider, however there are some issues receiving calls.
The INVITE looks like this:
INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 <http://sip:NCdWXsdSsaF2B@192.168.0.190:52284/> SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> ...
The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-)
Then Freeswitch does this of cause:
[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound
That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong?
participants (5)
-
Allan Kristensen
-
Andrew Cassidy
-
Ken Rice
-
Michael Jerris
-
Victor Chukalovskiy