[asterisk-users] Not able to register an Extension

Alonso Genis alonso at planetfone.com.br
Fri Nov 21 14:07:07 CST 2014


----- Mensagem original ----- 

> De: "akhilesh chand" <omakhileshchand at gmail.com>
> Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> 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 => <username>:<password>:<username>@<Sip Proxy IP or domain 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 < alonso at planetfone.com.br >
> wrote:

> > ----- Mensagem original -----
> 

> > > De: "akhilesh chand" < omakhileshchand at gmail.com >
> 
> > > Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> > > < asterisk-users at 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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> 
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> 

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> 
> > _____________________________________________________________________
> 
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> 
> > http://www.asterisk.org/hello
> 

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> 
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> 
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 

> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello

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