sip trunk issue

mutaz

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#1
i'm having problems configuring my sip trunk
i can make calls but i can't seem to receive any.
if i try my account on a softphone i can make and receive calls.
but when i configure it on elastix i can only make calls
where do you think the problem is?
 

Lee Sharp

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#2
Get on the console and type "asterisk -r" and make an inbound call with your cell. You will probably see some errors. That will help us figure out the problem.
 

fmvillares

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#3
in every sip trunk you have some basic settings common to all systems and some special ones for each one...we dont have the magic ball so please put some info debug etc to help us to help you...
 

albix

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#4
Hi to all,
I'm having the same problems configuring my sip trunk.
I'm using Elastix 2.0.3 64bit.

When I make an inbound call the output of my asterisk console is:

Code:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged on from 127.0.0.1
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Manager 'admin' logged off from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Could you help me?

Thanks in advance :)
albix
 

mutaz

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#5
i tried to check the concole but i dont get anything the cell phone just gives me disconnected when i call.
if i do the same thing with the softphone everything works fine
 

fmvillares

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#6
thats not enough but seems to be a codec issue or context.
 

mutaz

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fmvillares

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#8
do you have an inbound rule to let the calls get into the system by your did and get into an ivr or phone????????????
 

mutaz

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#9
sip
 

fmvillares

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#10
first of all
allow=g729&ilbc&ulaw&alaw ----this line is wrong....g729 is not applicable by default assuming you re a newbie since it requires licensing and payment bu channel use also ilbc due to licensing issue is not included in elastix by default...and the character & is not used in asterisk since the ages of 1.2 version...the only allowed separator is the comma
so a correct line would be allow=alaw,ulaw,gsm

User context: from-trunk
User details:
 

jiahau

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#11
I'm having the same issue. My outgoing trunk works but not incoming.

Here are my incoming trunk settings,

canreinvite=no
context=from-trunk
dissallow=all
allow=ulaw
fromuser=762994xx
insecure=very
qualify=no
secret=secret
type=friend
username=762994xx

Register String,
762994xx:secret@10.2.xx.xx/762994xx

Whenever I call the DID 762994xx, it only appears,
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

And from Wireshark that I managed to tap awhile ago, it shows that Elastix did receive INVITE from 10.2.xx.xx but replied it with a 404.

It's quite urgent now and I don't mind paying for consultant to help me as long as they can get the job done :(
Please help.
 

vascojuan1979

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#12
dissallow does not exist is disallow as a gramatic error the rest of the line get unprocessed
 

jiahau

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#13
Thanks and nice on spotting that vasco ;)

Unfortunately, that wasn't the solver :(

After some trial & error, I managed to get it working magically after I did 'service asterisk restart' based on this settings:

Incoming Settings
USER Context: from-trunk
USER Details:
host=10.2.xx.xx
type=friend
insecure=port,invite
dissallow=all
allow=ulaw
context=from-trunk

I have no idea why and seems like I only removed the authentication details (username, secret) and it worked. You might also noticed that the 'dissallow' wrong spelling was still there but it still worked, so I didn't change it. Anyhow, I quickly did a back-up from Elastix System Main page and happily called it a day.

Went over to client's site today and deployed it with the same settings. Strangely, it can't work now! Thinking maybe it could be due to my reboot and certain configs went haywire. So, I did a restore based on yesterday working configuration. It didn't help either.

I've tried changing to 'disallow' and did couple of 'service asterisk restart', didn't work either. Tried couple of settings and changes and restart, didn't work too. Also, I have the 'Allow Anonymous Inbound SIP Calls? = yes' on all the while.

Also, to test the trunks are fully functional, I register directly via ATA & 3CX softphone to the provider and they worked both ways. Somehow, with Elastix as the client, it can't accept incoming.

I'm really running out of ideas on how to solve this. I can't explain why it suddenly worked yesterday and not today even with the same configurations.

Could this be due to NAT? Since outgoing is fine and not incoming message?
My only chance tomorrow is to try editing sip.conf and play around with:

Externip=x.x.x.x
Localnet=x.x.x.x
Nat=yes

Appreciate any help I can get from anyone or any suggestion on what should I look out for. You can contact me directly via MSN at russel_loh@hotmail.com or Skype at russellx. We can work out a payment as long as my problem is solved.

Thank you.
 

fmvillares

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#14
u should try paid support as u dont have read the examples in elastix without tears or comunicaciones unificadas..
 

jiahau

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#15
FYI, I've read the EWT and have tried all the possible examples given from the trunking chapter.

By the way, due to my desperation 2 nights ago, I've also paid for 1hr 24/7 support but it took 7 hours to approve the payment and get my ticket #. During that time, I've already solved the problem but now it re-appears in different site.

That's why I'm suspecting could it be due to NAT-ed environment and I don't want to use the 1hr support first without trying to solve it myself ;)
 

fmvillares

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#16
if you re behind nat it will probably never work in a stable way..
u need direct internet connections with all the risks involved or simply vpns
 

netyco

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#17
Hello,

i have same problem :( i need solution for this? :(

thanks...
 

yotella

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#18
I cannot receive calls into my Elastix.

I cannot receive calls into my Elastix. Call fails with : == Using SIP RTP TOS bits 184


My Sip Client (Express Talk Business Edition soft phone) is connected to an Asterisk server (I now refer to it as Box A). When I make a call, it fails. The Soft phone logs as follows:
Code:
	19:25:37 Initiated sip call to: 102
	19:25:37 Error. Other side said: Service Unavailable
	19:25:37 Call has disconnected
In the orginating Asterisk server (the Box A) logs as follows:
Code:
 -- Executing [102@test:1] Dial("SIP/111-b7d04690", "SIP/102") in new stack
    -- Called 102
    -- SIP/102-08857e08 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/111-b7d04690' status is 'CONGESTION'
My Elastix (I call this BOX B) has a sip trunk that registers as 102 in BOX A. The above call only logs two lines on the cli:
Code:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
The issue is apparently caused by a bug as described here: http://bugs.elastix.org/view.php?id=245 .I updated my FreePBX to version 2.8.1.4 but I still get the same error when trying to get a call into my Box B (Elastix). This update done after trying 'yum update freePBX' on the console, a suggestion by Chilling_Silence here (http://bugs.elastix.org/bug_view_advanc ... bug_id=255) which did not solve my problem.

My freePBX modules are all updated to correspond with my current FreePBX 2.8.1, as indicated in Module Admin (https://<<Elastix IP>>/admin/config.php?type=setup&display=modules).


My Inbound route config is properly done. I have even tried to allow all calls in irrespective of caller id for a test but in vain. Also tried the insecure 'allow anonymous calls' in the General config.

How can I fix this issue?


Additional info on Box B (Elastix):
-----------------------------------


I am able to make calls without any problems from Box B (Elastix ) to all clients connected to Box A.

Code:
[root@sip2 ~]# rpm -qa elastix*
elastix-firstboot-2.0.0-13
elastix-email_admin-2.0.0-18
elastix-system-2.0.0-30
elastix-asterisk-sounds-1.2.3-1
elastix-2.0.0-36
elastix-vtigercrm-5.1.0-8
elastix-agenda-2.0.0-18
elastix-fax-2.0.0-12
elastix-reports-2.0.0-16
elastix-a2billing-1.3.0-4
elastix-addons-2.0.0-16
elastix-pbx-2.0.0-26
and

Code:
[root@sip2 ~]# rpm -qa aste*
asterisk-sounds-fr-1.6.2.10-1
asterisk-1.6.2.10-1
asterisk-perl-0.10-2
asterisk-addons-1.6.2.1-0
asterisk-sounds-es-1.6.2.10-1
asterisk-devel-1.6.2.10-1
Help!
 

yotella

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#19
I found my problem was the 'context' in the BOX B. Supposed to be 'context=from-trunk'. A silly mistake.
 

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