[asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

hi,

I have a really strange problem which is driving me crazy for days now.

If I register my asterisk (tried all versions from 1.6 up to 13.x) with 
one sip registrar,

everything works... calls go out and call come in... no 32 seconds limit.

but as soon as I configure another sip registration on another server, 
outgoing

calls  drop after 32 seconds.

as far as I know, there is no firewall in between...

I tried to "work around" this by increasing the settings for "timerb"... 
but I

realized that asterisk does not care at all, what I set this value to...
"sip show settings" always gives me 32000ms, and it does not make any
difference if I configure timerb in the general context or in the phone 
context...


any ideas?

thanks,
yves

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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Andreas Sikkema]

> but as soon as I configure another sip registration on another server,
> outgoing
> calls  drop after 32 seconds.

Are both your servers behind the same NAT router?

-- 
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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

but as soon as I configure another sip registration on another server,
outgoing
calls  drop after 32 seconds.

Are both your servers behind the same NAT router?


thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they 
interfere at all?


thanks,
yves

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Re: [asterisk-users] Not able to register an Extension

[akhilesh chand]

Hi Alonso,

Thanks for your reply but after setting the value of srvlookup=no i got
same error.


On Sat, Nov 22, 2014 at 1:37 AM, Alonso Genis 
wrote:

>
> - Mensagem original -
>
> > De: "akhilesh chand" 
> > Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Enviadas: Sexta-feira, 21 de novembro de 2014 16:54:35
> > Assunto: Re: [asterisk-users] Not able to register an Extension
>
> > Hi Alonso,
>
> > sip.conf
>
> > [general]
> > context=hunt_incoming
> > port=5060
> > bindaddr=0.0.0.0
> > srvlookup=yes
>
> Did you try to set srvlookup=no?
> http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup
>
> > disallow=all
> > allow=all
> > nat=yes
> > callerid = LITE
> > externip=
> > externhost=
> > autocreatepeer=yes
> > autodomain=yes
> > localnet=xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx
> > canreinvite=yes
> > language=En
> > allowtransfer=yes
> > realm=telunet
> > domain=192.168.1.5
> > maxexpiry=3600
> > defaultexpiry=200
> > useragent=LITE PBX
> > usereqphone = yes
> > dtmfmode = rfc2833
> > alwaysauthreject = no
> > regcontext=sipregistrations
>
> > rtptimeout=3600
> > rtpholdtimeout=300
> > rtcachefriends=yes
> > ;--- SIP DEBUGGING
> > ---
> > sipdebug = yes
> > registertimeout=60
> > registerattempts=5
> > callgroup=1
> > pickupgroup=1
> > callevents=yes
>
> > ;register => ::@ name>
>
> > [authentication]
>
> > [4001]
> > type=friend
> > context=outbound
> > defaultuser=4001
> > secret=4001
> > callerid="EXT1"
> > host=dynamic
> > nat=no
> > dtfmode=rfc2833
> > disallow=all
> > subscribecontext=outbound
> > canreinvite=no
> > allow=all
>
> > [4002]
> > type=friend
> > context=outbound
> > defaultuser=4002
> > secret=4002
> > callerid="EXT2"
> > host=dynamic
> > nat=no
> > dtfmode=rfc2833
> > disallow=all
> > subscribecontext=outbound
> > canreinvite=no
> > allow=all
>
> > [4003]
> > type=friend
> > context=outbound
> > defaultuser=4003
> > secret=4003
> > callerid="EXT3"
> > host=dynamic
> > nat=no
> > dtfmode=rfc2833
> > disallow=all
> > subscribecontext=outbound
> > canreinvite=no
> > allow=all
>
> > On Fri, Nov 21, 2014 at 11:52 PM, Alonso Genis <
> alo...@planetfone.com.br >
> > wrote:
>
> > > - Mensagem original -
> >
>
> > > > De: "akhilesh chand" < omakhileshch...@gmail.com >
> >
> > > > Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> > > > < asterisk-users@lists.digium.com >
> >
> > > > Enviadas: Sexta-feira, 21 de novembro de 2014 14:36:05
> >
> > > > Assunto: [asterisk-users] Not able to register an Extension
> >
>
> > > > Hi folk,
> >
>
> > > > I'm trying to register an extension through softphone and got
> stuck.I got
> >
> > > > below error:-
> >
>
> > > > [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via:
> > > > missing
> >
> > > > sent-by in Via header
> >
> > > > [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve:
> >
> > > > getaddrinfo("", "(null)", ...): Name or service not known
> >
> > > > [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could
> not
> >
> > > > resolve socket address for ''
> >
> > > > Sending to 192.168.1.2:5060 (NAT)
> >
> > > > [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via:
> > > > missing
> >
> > > > sent-by in Via header
> >
> > > > [Nov 22 07:31:28] ERROR[3522]: chan_sip.c:7897 process_via: error
> > > > processing
> >
> > > > via header
> >
> > > > [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:10420 respprep: error
> > > > processing
> >
> > > > via header, will send response to originating address
> >
>
> > > > Please let me know how could i solve the same and I will appreciate
> your
> >
> > > > suggestion.
> >
>
> > > Please, send us your sip.conf, i suspect is a problem with your
> bindaddr or
> > > name resolution.
> >
> > > Alonso.
> >
>
> > > > Thanks & Regards
> >
> > > > Akhilesh
> >
>
> > > > --
> >
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> >
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> >
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> >
>
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> >
> > > > To UNSUBSCRIBE or update options visit:
> >
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Eric Wieling]

Try setting directmedia=no in sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
>> but as soon as I configure another sip registration on another server,
>> outgoing
>> calls  drop after 32 seconds.
> Are both your servers behind the same NAT router?
>
thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they 
interfere at all?

thanks,
yves

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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Rafael Visser]

Hi Yves..
This may be silly... but what is the useragent of your sip configuration?
In the case that useragent has some special characters like "(.", please
remove it and tell us if there is any change!!.
Regards.
rv


2014-11-22 14:50 GMT-03:00 Eric Wieling :

> Try setting directmedia=no in sip.conf.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
> Sent: Saturday, November 22, 2014 8:06 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only
> when
>
> Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
> >> but as soon as I configure another sip registration on another server,
> >> outgoing
> >> calls  drop after 32 seconds.
> > Are both your servers behind the same NAT router?
> >
> thanks for taking part...
>
> I don´t know...
> one is
>
> siptrunk.ovh.net
>
> and the other one is
>
> sip.ovh.fr
>
> how can i determine and how could that affect... I mean... why do they
> interfere at all?
>
> thanks,
> yves
>
> ---
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>
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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Ron Wheeler]

You might check your phones as well.
We had this problem early on with a softphone and it was a setting in 
the phone that was set to hang up after 30 seconds of inactivity "in 
case of network disruption". For some reason it was detecting "network 
disruption" in every call even when the calls were proceeding normally.

Unchecking this box solved the problem.

It may not be related to your problem but if it is the cause, you will 
spend a lot of time trying to fix this in Asterisk. :-D At least I did!


On the bright side, it does force people to get point in a hurry!

Ron

On 22/11/2014 12:50 PM, Eric Wieling wrote:

Try setting directmedia=no in sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:

but as soon as I configure another sip registration on another server,
outgoing
calls  drop after 32 seconds.

Are both your servers behind the same NAT router?


thanks for taking part...

I don´t know...
one is

siptrunk.ovh.net

and the other one is

sip.ovh.fr

how can i determine and how could that affect... I mean... why do they
interfere at all?

thanks,
yves

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--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] High resident memory with 11.14.0 ?

[Freddi Hansen]

Its up to 5.8G of resident memory with 28321 calls processed.
The OOM killer is going to kill this soon at this rate (8GB RAM machine).
This seems like a pretty serious problem.
It looks like I'll need to restart asterisk every night
Hi the number of cpu cores that you see with top  times 512Mbyte is the 
level of ram that's needed


e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be 
( 2 x 8 x 2  x 0,5 gb ) = 16 gb  + a bit exstra.
So from start memory usage increases until it reaches 17.3 gb and then 
stabilizes. at that level.

You can disables hypertreading and cut your ram usage to half of that.

I can't see what hardware you are using but I think you need to check 
that the rule above fits your hardware.


b.r.
Freddi







On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna > wrote:

Hi,
I have an Asterisk server that's been running now for around 2 days.
I've noticed that the resident memory seems to be very high for its 
current call load:


  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+ COMMAND   
18321 asterisk  20   0 8050m 5.2g 6968 S   
13 66.2 363:11.80 asterisk


$ asterisk -rx "core show channels"

24 active channels

12 active calls

25216 calls processed


This server has a bunch of IAXModems hooked up to it and is mainly 
used as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory 
used after 2 days with only 12 currently active calls?


I am not using any realtime peers.

There are 100 registered SIP peers on this server as well.

Thanks.

-- James

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 4:57 PM, Larry Moore wrote:


On 23/10/2014 5:43 AM, Leandro Dardini wrote:

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);



What does your dialplan look like that makes the paging call?

Listing from my Asterisk:

'8000' => 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()


Using the Web Interface on you SPA501, log in as admin and select the
advanced view. Select the SIP tab, down the bottom of the page there is
a section headed 'Linksys Key System Parameters'.

You will want settings much like

Linksys Key System: yes
Multicast Address: 224.168.168.168:6061
Key System Auto Discovery: no
Key System IP Address: 
Force LAN Codec: 711a 


For the benefit of others I encountered a situation where I was getting 
one-way audio in a call regardless of it not being a paging call, this 
was because the negotiated codecs for the call was one other than the 
one selected in the 'Force LAN Codec:' setting.


It would appear setting the 'Force LAN Codec:' to either G711u or G711a 
_always_ enforces the phone to use this codec for its Encoder regardless 
of what is negotiated in SIP.


My advice, leave the 'Force LAN Codec:' setting at its default value 
which is 'none'.


Larry.


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