Messages

1

Features requests / Additional settings of SIP-registration

« on: July 19, 2016, 10:51:16 AM »

Hello colleagues!

Can you please tell me how to make additional registration settings on a operator's SIP-switch.

My Yate registred on a switch, simulating the SIP-gateways, with a set of different accounts.
According to the rules of this operator, one account at a time can make or receive only one connection.

Therefore, they require that lasts until the connection is established, our an account (which simulates the SIP-gateway) did not renew the SIP-registration. (Indeed, if it occurs, they SIP-switch breaks the existing telephone connection.)
Further, they are less confident voice explains that this is because the re-registration during the call, in the "Contact" field indicates another TCP port, than in the standby mode.

So I have two questions:
1. Can I customize Yate so that when the one connection is established an account did not renew the SIP-registration?
2. Can I customize Yate so specified at SIP-registration port not changed?

Ogogon.

2

Features requests / Re: How to collect the debugging information?

« on: March 10, 2016, 07:39:01 AM »

You can't filter debug based on sip account.

Regrettably.

You should stick with Wireshark (if possible) and yate log.
You should use yate log to see what happens when a connection is lost.

Unfortunately, nothing else to do.

And what do you mean by connection lost: sip call hangup or some TCP connection break?

I had to disable the remaining twenty SIP-accounts to get more or less readable log.
From the log of the account that was used as a test, it became clear that the operator suddenly has sent SIP-command "BYE".
Operator already have confirmed that the problem is on their side. Now they are all gone to think about why this is happening.

By the way, when instead Yate registered Asterisk, this does not happen.

Ogogon.

3

Features requests / Re: How to collect the debugging information?

« on: March 07, 2016, 09:24:16 AM »

Please checkout http://docs.yate.ro/wiki/Debugging_in_Yate

You probably need to turn on debug on SIP also ('debug sip level 10' command in telnet console).

Thank you. I have read this document for a long time.
Unfortunately, I did not find in it the answer to my question.

Maybe I was not careful, but I do not understand from him how to enable debugging only for one SIP system.

For example, Asterisk is done with a single command:
sip set debug peer peer_name

Is there any possibility at Yates?

Also, a wireshark capture can help.

Thank you. Sure, I can use an external sniffer, but I was hoping that Yate has a flexible functional of debugging.
External sniffer is not as good - for him, in my case, I need to write quite complex filters.
In addition, tсpdump and his relatives may lose a part of a package, but a built-in debugger - never.

Ogogon.

4

Features requests / How to collect the debugging information?

« on: March 04, 2016, 03:11:29 PM »

Dear colleagues, please tell me, how do I do the right thing in my case.

I have unexpected break of sessions, and I want to understand what the cause.

Architecture of constructed system is not entirely typical. This is caused by reasons that we can not change, and under who are forced to adapt.
We have an operator who does not want to take into consideration our technical specifics.
If support of CID is turn on, operator sends incoming invites in the RFC-3966 format, and softswitch Asterisk, that we use, does not understand this format. He does not accept these invites.
Now we are working with the CID off, but we need to turn it on.

Operator's Huawei <---> Asterisk

In order to get ready for the turn on of CID, we have created a transit softswitch. It Yates, who supports RFC-3966 format. He registers on the operator softswitch and transparently transmits calls to Asterisk.

Operator's Huawei <---> Yate <---> Asterisk

So we have an architecture that performs its task.

Unfortunately, after the iinclution of a transit unit in the form Yate, came an unexpected and unpleasant effect.
Sometimes, through different times, the connection is broken. Without Yayte this does not happen.
I want to collect debugging information and understand where it comes from a request for disconnection.

How to in Yate save SIP-session of selected SIP-account, or, better yet, all the debug information of sessions?


Advance grateful for the answer on the subject matter,
Ogogon.

5

Features requests / Re: How to manually set the argument for SIP method "REGISTER"?

« on: March 04, 2016, 01:19:07 PM »

Try to set registrar address in outbound parameter and registrar to required domain:

outbound=10.XXX.YYY.ZZZ:5060
registrar=ims.intranet.telephonic.com

Thank you. Your variant has successfully registered.

Ogogon.

6

Features requests / How to manually set the argument for SIP method "REGISTER"?

« on: March 03, 2016, 02:49:21 PM »

Dear colleagues, I have not quite typical question.

I need to manually set the argument for SIP method "REGISTER".

I use Yate for acces to multiple accounts of one large telephone operator.
Access is via a corporate network that uses the intranet address. DNS servers in it are not available.
Nevertheless, by their rules, the argument to the command "register" must be not IP-address, but the internal domain name of registration server. This is a condition of their authorization system and this issue is not discussed.

I set IP-address of the registrar, because appeal to their DNS is not possible:

registrar=10.XXX.YYY.ZZZ:5060

On this basis, Yate forms the argument of the method "REGISTER":

REGISTER sip:[10.XXX.YYY.ZZZ:5060] SIP/2.0

But according to the rules I need to create completely different argument.

REGISTER sip:ims.intranet.telephonic.com SIP/2.0

How can I do that?

Advance grateful for the answer on the subject matter,
Ogogon.

7

Yate users hangout place / Yate server status viewing

« on: June 28, 2015, 05:14:50 AM »

Hi, colleagues!

Please help me understand for themselves one fundamental position in work with the server Yate.

As I understand it, the Yate server not have console with command interpreter. In contrast to the Asterisk.
At Yate server has the ability to view the flow of log messages in the style of syslog.

Thus, at the time of starting or restarting the server I can see the status of all its units, modules, accounts, etc.

But runned communication server can not be restarted too often.

How can I see, for example, status of the server registration's to an external system without a restart?
Asterisk's analog - 'sip show registry'. Or other - 'sip show peers'.

Ogogon.

8

Other Yate server issues / IAX2 without listen on port 4569

« on: November 14, 2014, 09:37:12 PM »

Dear colleagues!

Is it possible to preserve the functionality of outgoing connections on IAX2, but disable listening on port 4569?

I do not needed a server IAX2, but I want to register as a IAX-client, to establish a trunk.

Ogogon.

9

Other Yate server issues / Re: Yate & RFC-2806

« on: November 14, 2014, 09:15:10 PM »

Thank you very much.

Ogogon.

10

Other Yate server issues / Re: Yate & RFC-2806

« on: November 14, 2014, 09:17:24 AM »

Hi,

Can you describe what exactly do you need?

Will Yate properly understood in the incoming invite fields without "sip:"?
Е.g.,

Code: [Select]

From: <tel:81234567890;cpc-rus=1;phone-context=+7>;tag=sbc1308j3izzjz8-CC-2
P-Called-Party-ID: <tel:+71234567890>
P-Asserted-Identity: <tel:81234567890;phone-context=+7>

11

Other Yate server issues / Yate & RFC-2806

« on: November 13, 2014, 06:59:05 PM »

Dear colleagues!

Tell me, please, whether the Yate has support of RFC-2806?
This additional schemes in SIP URI. For example: "tel:", "fax:", etc.

Unfortunately, provider's not opensource softswitch, on which I need to register, without these schemes can not. But the Asterisk from these schemes, falls into a riot.

Ogogon.