SIP call limit problem in Elastix 1.6.2-7

palillo

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#1
Hi,

I have a very strange problem. I have installed latest Elastix 1.6 X86_64 (see below for info about this installation and versions). I use Quintum Tenor 24 FXO port to connect to the PSTN.

The first thing I performed after installation was a yum update all to get latest version of all.

Asterisk 1.4.33.1
Elastix 1.6.2-7
Linux elastix 2.6.18-164.el5 #1 SMP Thu Sep 3 03:28:30 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux

Then, I manually updated to FreePBX 2.8.0.0 through the online module admin.

Apparently for an unknown reason, I am only able to make 2 outgoing simultanous calls through the Quintum or through any SIP device (for example, I can only make 2 calls from my eyebeam SIP softphone to the *97 voicemail app, the third attempt gets a SIP 488 not acceptable here message from Asterisk.

Looking into sip show inuse for the quintum trunk and for my extension (SIP/200), I found that max 2 peers (or 2 concurrent calls) are being allowed. When trying to make a third call, you get SIP 488 NOT ACCEPTABLE HERE from Asterisk.

*CLI> sip show inuse
* Peer name In use Limit
quintum2 2/1 50
200 2/0 50

After reaching 2/1 or 2/0, no more calls are able to be made from that SIP device or to the outside world through the quintum2 PSTN gateway.

I have verified the call-limit is set to 50 on all configurations and you can see this on the "sip show inuse" command.

I have also found that sip_general_additional.conf is bundled with a apparently bad limitonpeer command because it was ended with an "s" (limitonpeers). I tried with limitonpeer and limitonpeers with yes/no with the same result.

How this system was installed...

First of all, this was a migration from an very old Trixbox.

1. First, we upgraded the FreePBX of the Trixbox old install to 2.8.0.0 to be able to perform a backup.
2. Second, we updated the new FreePBX on the Elastix server to 2.8.0.0 to be able to restore from Trixbox conf.
3. Third, we copied all audio files from Trixbox to Elastix (to have all custom made audios for the IVR, etc).
4. Then I copied the /var/lib/asterisk/astdb so the Asterisk DB is the same for registration and all that things.
5. Fifth, I fixed the /etc/asterisk/manager.conf so FreePBX can connect to Asterisk through AMI (remember it was restored from Trixbox backup with wrong username/password).
6. I fixed username/password in /etc/asterisk/cdr_mysql.conf as this was restored incorrectly from Trixbox backup.

After all the above, I have a working Asterisk that can call between extensions, have working IVRs, trunks to the PSTN, extensions, IAX2 trunks to external Asterisks and all FreePBX settings and configuration in all modules.

Apparently all is working fine!!!

The problem I have is the limit on calls as only 2 calls can be made through any SIP device.

I have either downgraded to Asterisk 1.4.31-2 according to a recent problem I found on Xorcom website related to DAHDI. I know I am not using DAHDI (only SIP/IAX2) but who knows.... This downgrade did not fix my problem.

Below you can find the sip.conf sip_general_additional.conf for quintum2 SIP trunk and SIP/200 extension. Nothing strange here. This configuration have been working for a couple of years on Trixbox before updating without a problem.

vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
language=es
jbenable=no
srvlookup=no
maxexpiry=3600
minexpiry=60
defaultexpiry=120
allowguest=yes
registerattempts=0
maxcallbitrate=384
registertimeout=20
notifyhold=yes
g726nonstandard=no
t38pt_udptl=no
videosupport=no
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
notifyringing=yes
checkmwi=10
rtpkeepalive=0
nat=yes
externip=xxxxxxxxxxxx
localnet=192.168.1.0/255.255.255.0
localnet=10.0.10.0/255.255.255.0

[quintum2]
disallow=all
username=xxxxxxx
type=friend
secret=xxxxx
insecure=port,invite
host=192.168.1.249
context=from-pstn
nat=no
allow=alaw
allow=ulaw
qualify=yes

[200]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=xxxxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
allow=g729
dial=SIP/200
mailbox=200@default
permit=0.0.0.0/0.0.0.0
callerid=device <200>
call-limit=50

Any comments or help on this is appreciated.

If you need further information, please let me know.

As a last note, I am able to make 6 simoultaneus calls from my eyebeam on my Asterisk 1.4.29 Elastix 1.6 at my office without any problem.

Best regards.

Andres Maduro
 

jessie

Joined
Sep 17, 2008
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#2
Hi Andres Maduro,

The problem is not in your Elastix, it's in your Quintum. Remember, you have to configure your Quintum channel individually. Each channel should acts like an extension. So if you configure only two channel in the Quintum, it will definitely gives you two lines only.


Cheers,


Jessie
 

jessie

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Sep 17, 2008
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#3
Hi Andres Maduro,

The problem is not in your Elastix, it's in your Quintum. Remember, you have to configure your Quintum channel individually. Each channel should acts like an extension. So if you configure only two channel in the Quintum, it will definitely gives you two lines only.


Cheers,


Jessie
 

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