Topic: SAIL - Sipgate incoming call problems

[mark_s_tt]

Hi all,

I've got Sail up and working, and I like it a lot. Problem is I'm struggling with incoming calls.

Extension to extension calls, as well as outbound are working fine. Carrier is Sipgate.co.uk and thanks to Selintra, setting up was no problem as they were already in the carriers list.

Problem is incomming calls, as I said, Sail is registering with Sipgate fine, and I can see the calls coming in with sip debug turned on in the asterisk console but am getting a 404 not found for every incoming call.

Here's the debug info (or at least the only part with a problem:

<--- Reliably Transmitting (no NAT) to 217.10.79.23:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bKc8aa.ed67f964.0;received=217.10.79.23
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKc8aa.b5d2e733.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK31d9d49c;rport=5060
From: "Unknown" <sip:Unknown@217.10.66.71>;tag=as346a5817
To: <sip:448458692823@217.10.79.8>;tag=as2c9b8e9f
Call-ID: 377d497b2e98252972c9945c0e65accd@217.10.66.71
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


Sidenote, the Unknown is because I tried from an ex-directory number, if I try from my mobile, the number shows up but the problem is the same.

Been trying all sorts all day, so thought I'd post and see if anyone had any ideas.

Thanks a lot.

[SARK devs]

HI there,

First things first.  SIPGATE delivers the DND (dialed number) in E164 format.  Have you correctly created your sipgate account with the number as E164?

E164 is  Countrycode/areacode/subscribernumber.   Thus a UK London (0207) number would look like 442071934840.  

If you haven't created your SipGate Trunk DiD in this way when you put the account up, don't worry.  Just create a free-standing PTT-DID with the correct number (or delete the SipGate Trunk and redo it).

If you want to read up on SIP carriers and how Asterisk talks to them, there is a nice little write-up in the docs here

http://switch/docs/cgi-bin/view/Main/DocChapter259


Kind Regards

S

[mark_s_tt]

Hi Selintra,

Thank you for the response. I tried both, first I added an E164 DID in the PTT_DID_Group panel, this still didn't work so I deleted the trunk all together, and created a new one using the E164 number.

I also added a non E164 DID in the PTT_DID_Group panel for good measure with the new trunk after the new trunk still didn't work.

Can still make outbound calls on the new trunk, just inbound fail. 404 message exactly the same.

Thanks
Mark.

[SARK devs]

HI Mark,

I had a quick look at the Sipgate website and the Asterisk SIP setup seems to suggest they are delivering against the account user-id, not the dialled number (quite a few SIP carriers do this - we've never really understood why).

In extensions.conf they create an extension as follows....

exten => {seven digit SIP-ID},1,Dial(SIP/internestelefon,20,tr)

So....

Create a PTT-DID using your useraccount number instead of the telephone number (all a PTT-DID actually does is to create an extension in extensions.conf for an incoming SIP call to anchor to - asterisk does the rest).

With a little bit of luck that should do it for you.



Best

S

[mark_s_tt]

Hi Selintra,

It Worked!!!

I must admit, considering you give support for your product to the SME community by choice, and at this level, really is quite amazing.

Thank you, thank you, thank you.

I would have never clicked this was the problem, so without your help I doubt I'd have got this working.

Kindest regards
Mark.

Right, now I can get stuck in!