[asterisk-users] asterisk-users Digest, Vol 73, Issue 5
Nasir Javaid
nasirjavaidnasir at gmail.com
Tue Aug 3 10:22:17 CDT 2010
Hi C F
no asterisk and sip device are not behind same router. actually both are in
different countries. how ever when caller and callee are behind same routers
voice is just fine (both ways) and i can see re-INVITEs too.
but when someone calls from another router then this issue arises. caller
can hear the called party but called party can not hear caller. and there
are no re-invites issued too.
i am bit new to sip and rtp stuff and don't know what is going on. how
asterisk is issuing re-invites for devices behind same router and not for
device behind another router?
Nasir Javaid
Message: 12
> Date: Tue, 3 Aug 2010 07:21:06 -0400
> From: C F <shmaltz at gmail.com>
> Subject: Re: [asterisk-users] RTP stream not passing through router
> with port forwarding
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTin9G14ipFL3yVMsFQMtiY=b9Wgfci4XeRDRBDPu at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Is asterisk and the SIP device behind the same router?
> Most routers will not redirect internal NAT requests. So that if you
> are trying to have port forwarding done but the request and the
> forwarding destination are on the same interface it won't work.
>
> On 8/3/10, Nasir Javaid <nasirjavaidnasir at gmail.com> wrote:
> > Hi,
> >
> > I am trying to dial a registered user via his IP:Port mechanism, but
> problem
> > is that the audio data is not reaching to dialed user. here is the
> scenario.
> >
> > caller and callee both are registered at asterisk server. asterisk server
> is
> > on public ip so no port forwarding and natting necessary there. however
> > caller and callee both are behind router and there is port forwarding
> > enabled and nat=yes, qualify=yes in sip.conf for both users.
> >
> > callee user name: adf
> > callee local ip/port: 192.168.0.10:5678
> > callee router ip: 116.79.x.x
> >
> > when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly
> > fine to 192.168.0.10 through router and INVITE is sent to local ip
> through
> > router.
> >
> > INVITE sip:adf at 192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to
> > contact local ip through router and sends rtp there)
> >
> > but problem arises when i dial using IP:Port combination like this
> >
> > Dial(SIP/adf at 116.79.x.x:5678)
> >
> > In this case INVITE is sent to router ip instead of local ip through
> router.
> >
> > INVITE sip:adf at 116.79.x.x:5678 SIP/2.0 (asterisk sends rtp to router
> ip
> > and not local ip)
> >
> > Similerly TO header also has same ip as INVITE. I think in IP dial rtp is
> > not reaching to local ip through router as INVTE is meant for router ip
> and
> > asterisk does not know where to send rtp stream after sending it to
> router.
> >
> > how can this issue be resolved? is there something to be done at router
> > confiurations or sip.conf parameters. I have already played with
> > nat/qualify/canreinvite/directrtp/externip etc parameters.
> >
> > regards,
> >
> > Nasir Javaid
> >
>
>
>
> ------------------------------
>
> Message: 13
> Date: Tue, 03 Aug 2010 13:21:23 +0200
> From: Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de>
> Subject: Re: [asterisk-users] mapping of disconnect reasons
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <4C5817D3.976.37D9D61 at klitzing.pool.informatik.rwth-aachen.de>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi!
>
> > Is there a way to change the mappings of disconnect reasons to certain
> > SIP messages? E.G. I need to change the mapping for SIP 402 "Payment
> > Required" from 16 (normal termination) like it is in 1.4.24 to 21
> > (call rejected) as defined in RFC 3398.
>
> * if you think the mapping is wrong, then you should open a ticket on the
> Asterisk bug tracker
>
> * the mapping can only be changed in the code - which you ahve
>
> * Asterisk 1.8 will allow to read SIP response codes in the dialplan via
> {HASH(SIP_CAUSE,<channel-name>)}. Asterisk 1.8 also comes with a
> 'use_q850_reason' configuration option for generating and parsing, if
> available, "Reason: Q.850;cause=<cause code>".
>
> Philipp
>
>
>
>
> ------------------------------
>
> Message: 14
> Date: Tue, 03 Aug 2010 13:21:23 +0200
> From: Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de>
> Subject: Re: [asterisk-users] Codec negotiation : expecting G726,
> getting G711a (alaw)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <4C5817D3.31146.37D9EE8 at klitzing.pool.informatik.rwth-aachen.de>
> Content-Type: text/plain; charset=US-ASCII
>
> Hi!
>
> > Question 1 :
> > [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
> > audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
> > why is combined alaw|g726 and not g726|alaw (reverse) ??
>
> Guess: Here the order presented has no meaning for the order of codec
> negotiation.
>
> > Question 2 :
> > why do I see on my Grandstream phone that the codec being used is alaw in
> > stead of g726 ??
>
> Because that is what the phone and Asterisk have negotiated. ;-)
>
> > Question 3 :
> > How can I get g726 as first preferred codec ??
>
> Which Asterisk version are you using?
>
> * check if you have disallow/allow settings in the [general] section of
> sip.conf. Depending on your Asterisk version only the order in [general]
> would be respected, but not the order in the individual sip peer/user
> definition
>
> * look at the variable SIP_CODEC for the inbound (first) call leg, and in
> Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
>
> * many Asterisk operators have applied the third party "codec negotiation
> patch"
>
> Philipp
>
>
>
>
> ------------------------------
>
> Message: 15
> Date: Tue, 3 Aug 2010 07:26:41 -0400
> From: C F <shmaltz at gmail.com>
> Subject: Re: [asterisk-users] Caller ID issue
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTi=S6fboEqySvPVw25TmEvKDPNjbvjmsvniWuhbS at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> In most cases wait(.5) will do. I would not recommend using
> answer(2000) as that answers the channel, which means you start
> getting billed.
>
> On 8/2/10, Peder <peder at networkoblivion.com> wrote:
> >> I am using T1's and didn't think the spill would take that long.
> >
> >> PRI no, E&M yes.
> >
> > Some PRI take that long too because the telco sends the name in a
> followup
> > message, not in the initial call setup.
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> ------------------------------
>
> Message: 16
> Date: Tue, 03 Aug 2010 07:51:51 -0400
> From: John Novack <jnovack at stromberg-carlson.org>
> Subject: Re: [asterisk-users] chinaroby fxo card - never heard of them
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C5802D7.3080800 at stromberg-carlson.org>
> Content-Type: text/plain; charset="iso-8859-1"
>
> They seem to have taken over manufacture of cards Digium has discontinued.
> I have used several of the TE110 card with success and they are identical
>
> John Novack
>
>
> asteriskguru asteriskguru wrote:
> > hi,
> > I am using this card and IP phone about 6 months. There is no issues
> > at all.
> >
> > Installation procedures are same as Digium analog card.
> >
> > Hope it helps,
> > Ashik
> >
> > On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy <landysaccount at yahoo.com
> > <mailto:landysaccount at yahoo.com>> wrote:
> >
> > Hello.
> >
> > I'm looking to buy a FXO card to do some testing with two phone
> > lines I have at home and was looking in ebay some and found some
> > cheap ones but, the I've never heard of the brand or manufacturer:
> > chinaroby. They run for about $99 plus shipping. Have any one used
> > these? or please recommend one... Money IS an issue.
> >
> > Thanks.
> >
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> --
>
> Dog is my Co-pilot
>
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> ------------------------------
>
> Message: 17
> Date: Tue, 3 Aug 2010 09:11:07 -0400
> From: Zeeshan Zakaria <zishanov at gmail.com>
> Subject: Re: [asterisk-users] RTP stream not passing through router
> with port forwarding
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTi=GpvmFhHkAXHFP6BJgRUsVp6pXNz02WN1MjxBH at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> We all know what are you trying to do, and it is not possible to do, but it
> is very impolite and annoying to repost it every few days as a new post
> with
> a slightly different subject. Nobody else does it, and you too please avoid
> doing it.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-08-03 7:35 AM, "C F" <shmaltz at gmail.com> wrote:
>
> Is asterisk and the SIP device behind the same router?
> Most routers will not redirect internal NAT requests. So that if you
> are trying to have port forwarding done but the request and the
> forwarding destination are on the same interface it won't work.
>
>
> On 8/3/10, Nasir Javaid <nasirjavaidnasir at gmail.com> wrote:
> > Hi,
> >
> > I am trying to dial a registe...
> --
> _____________________________________________________________________
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> ------------------------------
>
> Message: 18
> Date: Tue, 03 Aug 2010 15:15:33 +0200
> From: Jonas Kellens <jonas.kellens at telenet.be>
> Subject: Re: [asterisk-users] Codec negotiation : expecting G726,
> getting G711a (alaw)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C581675.6000708 at telenet.be>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello Philipp,
>
> thank you for your answer.
>
>
> On 08/03/2010 01:21 PM, Philipp von Klitzing wrote:
> >> Question 3 :
> >> How can I get g726 as first preferred codec ??
> >>
> > Which Asterisk version are you using?
> >
>
> Using Asterisk 1.4.30
>
> > * check if you have disallow/allow settings in the [general] section of
> > sip.conf. Depending on your Asterisk version only the order in [general]
> > would be respected, but not the order in the individual sip peer/user
> > definition
> >
>
> In the [general] section of sip.conf I have :
>
> disallow=all
> allow=g726
> allow=alaw
> allow=g729
> allow=gsm
>
> > * look at the variable SIP_CODEC for the inbound (first) call leg, and in
> > Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
> >
>
> When I read the value of this variable just before the Dial()-statement,
> it is empty.
>
>
>
> Jonas.
>
>
>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
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> End of asterisk-users Digest, Vol 73, Issue 5
> *********************************************
>
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