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Mike Machado

2004-Jan-04 21:16 UTC

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:

WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
seqno 0 (Response)

When doing traces with ethereal, I see successful SIP and SDP
handshakes, but when * sends handytone RTP packets, I see a ICMP Port
Unreachable messages sent from Handytone to * regarding the UDP RTP
packet. * then gives up and I see a BYE from *, which handytone acks.

Handytone config is default except obvious SIP registration parameters.
I also have a Sipura SPA2000 and everything works perfect for that one,
same extension and everything (not at same time of course).

sip.conf entry:

disallow=all                    ; Disallow all codecs
allow=ilbc
allow=ulaw                      ; Allow codecs in order of preference

[131]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=300
callerid="handytone <131>"
mailbox=131
nat=0


Handytone info:

Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
HTML--1.0.0.19


Both on same subnet, no NAT. I have two Handytones, both exhibit same
symptoms. 

Anyone else have this problem?


--

Masakazu Nakano

2004-Jan-04 21:42 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

Hi Mike

I know exacty same situation about BT100 that sometimes lost any packets.

like a DoS attack for BT100? ;-(

mack_jpn

[hogehoge@mack asterisk]# ping 192.168.XX.XX
PING 192.168.XX.XX (192.168.XX.XX) from 192.168.XX.X : 56(84) bytes of
data.
64 bytes from 192.168.XX.XX: icmp_seq=0 ttl=250 time=2 usec
Warning: time of day goes back, taking countermeasures.
64 bytes from 192.168.XX.XX: icmp_seq=1 ttl=250 time=969 usec
64 bytes from 192.168.XX.XX: icmp_seq=2 ttl=250 time=766 usec
64 bytes from 192.168.XX.XX: icmp_seq=3 ttl=250 time=746 usec
64 bytes from 192.168.XX.XX: icmp_seq=4 ttl=250 time=829 usec
64 bytes from 192.168.XX.XX: icmp_seq=5 ttl=250 time=725 usec
64 bytes from 192.168.XX.XX: icmp_seq=6 ttl=250 time=735 usec
64 bytes from 192.168.XX.XX: icmp_seq=7 ttl=250 time=703 usec
64 bytes from 192.168.XX.XX: icmp_seq=9 ttl=250 time=670 usec
64 bytes from 192.168.XX.XX: icmp_seq=10 ttl=250 time=728 usec
64 bytes from 192.168.XX.XX: icmp_seq=11 ttl=250 time=711 usec
64 bytes from 192.168.XX.XX: icmp_seq=12 ttl=250 time=701 usec
64 bytes from 192.168.XX.XX: icmp_seq=13 ttl=250 time=707 usec
64 bytes from 192.168.XX.XX: icmp_seq=14 ttl=250 time=693 usec
64 bytes from 192.168.XX.XX: icmp_seq=15 ttl=250 time=692 usec
64 bytes from 192.168.XX.XX: icmp_seq=16 ttl=250 time=678 usec
64 bytes from 192.168.XX.XX: icmp_seq=17 ttl=250 time=673 usec
64 bytes from 192.168.XX.XX: icmp_seq=18 ttl=250 time=699 usec
64 bytes from 192.168.XX.XX: icmp_seq=19 ttl=250 time=683 usec
64 bytes from 192.168.XX.XX: icmp_seq=20 ttl=250 time=696 usec
64 bytes from 192.168.XX.XX: icmp_seq=21 ttl=250 time=714 usec
64 bytes from 192.168.XX.XX: icmp_seq=22 ttl=250 time=704 usec
64 bytes from 192.168.XX.XX: icmp_seq=23 ttl=250 time=701 usec
64 bytes from 192.168.XX.XX: icmp_seq=24 ttl=250 time=691 usec
64 bytes from 192.168.XX.XX: icmp_seq=25 ttl=250 time=670 usec
64 bytes from 192.168.XX.XX: icmp_seq=26 ttl=250 time=690 usec
64 bytes from 192.168.XX.XX: icmp_seq=27 ttl=250 time=698 usec
64 bytes from 192.168.XX.XX: icmp_seq=28 ttl=250 time=713 usec
64 bytes from 192.168.XX.XX: icmp_seq=29 ttl=250 time=723 usec
64 bytes from 192.168.XX.XX: icmp_seq=30 ttl=250 time=703 usec
64 bytes from 192.168.XX.XX: icmp_seq=31 ttl=250 time=694 usec
64 bytes from 192.168.XX.XX: icmp_seq=32 ttl=250 time=685 usec
64 bytes from 192.168.XX.XX: icmp_seq=33 ttl=250 time=727 usec
64 bytes from 192.168.XX.XX: icmp_seq=34 ttl=250 time=720 usec
64 bytes from 192.168.XX.XX: icmp_seq=37 ttl=250 time=687 usec
64 bytes from 192.168.XX.XX: icmp_seq=38 ttl=250 time=704 usec
64 bytes from 192.168.XX.XX: icmp_seq=39 ttl=250 time=686 usec

--- 192.168.XX.XX ping statistics ---
40 packets transmitted, 37 packets received, 7% packet loss
round-trip min/avg/max/mdev = 0.002/0.695/0.969/0.126 ms

On Sun, 04 Jan 2004 20:16:31 -0800
Mike Machado <mike@homelandtel.com> wrote:
> I am trying to get the handytone 286 to make a very simple call to * and
> having problems. It registers with * just fine, but when I place a call
> (to echo test, for example), the RTP stream seems to have problems
> opening. Here is there error I get in *:snip

John Baker

2004-Jan-04 21:55 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

I had a similar problem with a Cisco phone, i.e., the "Maximum retries
exceeded on call" error.
It took three days to track down the error to buggy network hardware.

Same symptoms, too - phone registered, one way conversation was ok (had a
test extension
for music on hold)

Fixed the hardware, phone works great.

John

----- Original Message ----- 
From: "Mike Machado" <mike@homelandtel.com>
To: <asterisk-users@lists.digium.com>
Sent: Sunday, January 04, 2004 10:16 PM
Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems

> I am trying to get the handytone 286 to make a very simple call to * and
> having problems. It registers with * just fine, but when I place a call
> (to echo test, for example), the RTP stream seems to have problems
> opening. Here is there error I get in *:
>
> WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
> exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
> seqno 0 (Response)
>
> When doing traces with ethereal, I see successful SIP and SDP
> handshakes, but when * sends handytone RTP packets, I see a ICMP Port
> Unreachable messages sent from Handytone to * regarding the UDP RTP
> packet. * then gives up and I see a BYE from *, which handytone acks.
>
> Handytone config is default except obvious SIP registration parameters.
> I also have a Sipura SPA2000 and everything works perfect for that one,
> same extension and everything (not at same time of course).
>
> sip.conf entry:
>
> disallow=all                    ; Disallow all codecs
> allow=ilbc
> allow=ulaw                      ; Allow codecs in order of preference
>
> [131]
> type=friend
> host=dynamic
> reinvite=no
> canreinvite=no
> qualify=300
> callerid="handytone <131>"
> mailbox=131
> nat=0
>
>
> Handytone info:
>
> Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
> HTML--1.0.0.19
>
>
> Both on same subnet, no NAT. I have two Handytones, both exhibit same
> symptoms.
>
> Anyone else have this problem?
>
>
> -- 
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Mike Machado

2004-Jan-04 23:17 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

I guess that was another thing that was strange. When I talked, I saw no
RTP coming from the handytone to *. Would there be a reason the
handytone would not send RTP until it successfully received a RTP packet
from *, but since its not accepting RTP, it would not send it either? 

I do not even get one way communication, I get no way communication.


Does anyone out there have this firmware version of Handytone working at
all with *?

On Sun, 2004-01-04 at 20:55, John Baker wrote:> I had a similar problem with a Cisco phone, i.e., the "Maximum retries
> exceeded on call" error.
> It took three days to track down the error to buggy network hardware.
> 
> Same symptoms, too - phone registered, one way conversation was ok (had a
> test extension
> for music on hold)
> 
> Fixed the hardware, phone works great.
> 
> John
> 
> ----- Original Message ----- 
> From: "Mike Machado" <mike@homelandtel.com>
> To: <asterisk-users@lists.digium.com>
> Sent: Sunday, January 04, 2004 10:16 PM
> Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
> 
> 
> > I am trying to get the handytone 286 to make a very simple call to *
and
> > having problems. It registers with * just fine, but when I place a
call
> > (to echo test, for example), the RTP stream seems to have problems
> > opening. Here is there error I get in *:
> >
> > WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum
retries
> > exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
> > seqno 0 (Response)
> >
> > When doing traces with ethereal, I see successful SIP and SDP
> > handshakes, but when * sends handytone RTP packets, I see a ICMP Port
> > Unreachable messages sent from Handytone to * regarding the UDP RTP
> > packet. * then gives up and I see a BYE from *, which handytone acks.
> >
> > Handytone config is default except obvious SIP registration
parameters.
> > I also have a Sipura SPA2000 and everything works perfect for that
one,
> > same extension and everything (not at same time of course).
> >
> > sip.conf entry:
> >
> > disallow=all                    ; Disallow all codecs
> > allow=ilbc
> > allow=ulaw                      ; Allow codecs in order of preference
> >
> > [131]
> > type=friend
> > host=dynamic
> > reinvite=no
> > canreinvite=no
> > qualify=300
> > callerid="handytone <131>"
> > mailbox=131
> > nat=0
> >
> >
> > Handytone info:
> >
> > Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
> > HTML--1.0.0.19
> >
> >
> > Both on same subnet, no NAT. I have two Handytones, both exhibit same
> > symptoms.
> >
> > Anyone else have this problem?
> >
> >
> > -- 
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users--

2004-Jan-04 23:21 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

Hi Mike,

I have handytones working OK with *.

My username and name of the context set to same.

Try this;

[131]
username=131
type=friend
host=dynamic
disallow=all
allow=alaw
allow=ulaw

However I am wondering why you get destination unreacherbale from the
handytone. This is nothing to do with SIP negotiation. So you might want to
go look at the RTP trace and see whether those ports are blocked in your
linux box. SIPURA might be using different set of ports.

Also you should upgrade the handytones to a better code. I have 1.0.4.26,
which is known to be pretty stable. If u want the code, search the mailing
list, I remember in December someone posted where to download the code.

SW


Message: 1
From: Mike Machado <mike@homelandtel.com>
To: asterisk-users@lists.digium.com
Date: Sun, 04 Jan 2004 20:16:31 -0800
Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
Reply-To: asterisk-users@lists.digium.com

I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:

WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
seqno 0 (Response)

When doing traces with ethereal, I see successful SIP and SDP
handshakes, but when * sends handytone RTP packets, I see a ICMP Port
Unreachable messages sent from Handytone to * regarding the UDP RTP
packet. * then gives up and I see a BYE from *, which handytone acks.

Handytone config is default except obvious SIP registration parameters.
I also have a Sipura SPA2000 and everything works perfect for that one,
same extension and everything (not at same time of course).

sip.conf entry:

disallow=all                    ; Disallow all codecs
allow=ilbc
allow=ulaw                      ; Allow codecs in order of preference

[131]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=300
callerid="handytone <131>"
mailbox=131
nat=0


Handytone info:

Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
HTML--1.0.0.19


Both on same subnet, no NAT. I have two Handytones, both exhibit same
symptoms.

Anyone else have this problem?

robert ivanc

2004-Jan-05 05:40 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

Mike Machado wrote:
>I guess that was another thing that was strange. When I talked, I saw no
>RTP coming from the handytone to *. Would there be a reason the
>handytone would not send RTP until it successfully received a RTP packet
>from *, but since its not accepting RTP, it would not send it either? 
>
>I do not even get one way communication, I get no way communication.
>
>
>Does anyone out there have this firmware version of Handytone working at
>all with *?
>  
>it works ok for me, for the most part. same firmware.

regards,
          robert

Mike Machado

2004-Jan-05 20:24 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems - FIXED

For the benefit of the archives, here is what I did to fix the problem:

Simply reordering

allow=ilbc
allow=ulaw

to 

allow=ulaw
allow=ilbc

in sip.conf

Fixed the problem. I guess the Handytone is not that graceful in codec
negotiation?

I did notice * sends RTP packets with payload type 97, which ethereal
says is an unknown type. * also sends the payload 97 to sipura, but
after a few packets, it starts to send ones with payload type 97, which
ethereal identifies as "ITU-T G.711 PCMU", maybe after it notices
sipura
is sending them payload type 0 packets. In the case of handytone,
handytone sends NO RTP packets to *, so maybe it does not get the clue.



On Sun, 2004-01-04 at 20:16, Mike Machado wrote:> I am trying to get the handytone 286 to make a very simple call to * and
> having problems. It registers with * just fine, but when I place a call
> (to echo test, for example), the RTP stream seems to have problems
> opening. Here is there error I get in *:
> 
> WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
> exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
> seqno 0 (Response)
> 
> When doing traces with ethereal, I see successful SIP and SDP
> handshakes, but when * sends handytone RTP packets, I see a ICMP Port
> Unreachable messages sent from Handytone to * regarding the UDP RTP
> packet. * then gives up and I see a BYE from *, which handytone acks.
> 
> Handytone config is default except obvious SIP registration parameters.
> I also have a Sipura SPA2000 and everything works perfect for that one,
> same extension and everything (not at same time of course).
> 
> sip.conf entry:
> 
> disallow=all                    ; Disallow all codecs
> allow=ilbc
> allow=ulaw                      ; Allow codecs in order of preference
> 
> [131]
> type=friend
> host=dynamic
> reinvite=no
> canreinvite=no
> qualify=300
> callerid="handytone <131>"
> mailbox=131
> nat=0
> 
> 
> Handytone info:
> 
> Software Version:    Program--1.0.4.17    Bootloader--1.0.0.11
> HTML--1.0.0.19
> 
> 
> Both on same subnet, no NAT. I have two Handytones, both exhibit same
> symptoms. 
> 
> Anyone else have this problem?--

Matteo Brancaleoni

2004-Jan-07 02:50 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

Hi.

I have the same issue with budgetones 102 (& 101) with firmware 1.0.4.30
But happens also with .4.26 , .4.18 and .4.17 .
Doing an ethereal trace, I noticed that the GS isn't answering to OK's
sent by asterisk when the ringed party answers (GS doesn't not send ACK
to the cpnnection confirmation), so after x seconds (6, more or less)
asterisk closes up the connection and so you get iCMP unreacheable
on the RTP port (that's ok since * closed the port).

So GS must fix that... send ACK to the 200/OK of the connection
confirmation.

Pretty interesting is that when you call to another * channel that's not
SIP (like ZAP,CAPI) all works ok... only SIP<->SIP raises the problem,
or better only when the SIP call is initiated by the GS to another SIP
device. If the call is started from a cisco to the GS, all works ok.

Matteo.

Il lun, 2004-01-05 alle 05:16, Mike Machado ha scritto:> I am trying to get the handytone 286 to make a very simple call to * and
> having problems. It registers with * just fine, but when I place a call
> (to echo test, for example), the RTP stream seems to have problems
> opening. Here is there error I get in *:
> 
> WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
> exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
> seqno 0 (Response)
> 
> When doing traces with ethereal, I see successful SIP and SDP
> handshakes, but when * sends handytone RTP packets, I see a ICMP Port
> Unreachable messages sent from Handytone to * regarding the UDP RTP
> packet. * then gives up and I see a BYE from *, which handytone acks.
> 
> Handytone config is default except obvious SIP registration parameters.
> I also have a Sipura SPA2000 and everything works perfect for that one,
> same extension and everything (not at same time of course).
> 
> 
> Both on same subnet, no NAT. I have two Handytones, both exhibit same
> symptoms. 
> 
> Anyone else have this problem?-- 
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web   : http://www.espia.it
Phone : +39 02 70633354      - ext 201
IAX(2): guest@213.140.14.155 - ext 201
Iaxtel: 1-700-56-62458       - ext 201

TeleSIP

2004-Jan-07 08:24 UTC

head link

[Asterisk-Users] Grandstream Handytone 286 RTP Problems

Hi Matteo,

I see a problem here.  And its the same as the trace I examined from
Wipeout.

After * sends back the "183 Session Progress" message, it should also
send
the "STATUS 200 OK" once the call is answered (the only STATUS 200 OK
I see
is the response to an INFO Message).  Since the GS never receives this
"STATUS 200 OK", it never sends back the ACK and the call will have a
"choppy" sound.

On the other file (cisco to GS), the "STATUS 200 OK" is clearly there
and
shortly after you can see the ACK

Try to find out where the STATUS 200 OK is being lost.

Look at page #5 of the attached PDF to see that the STATUS 200 OK message is
indeed needed

Regards,
Andres.
http://www.telesip.net

----- Original Message ----- 
From: "Matteo Brancaleoni" <mbrancaleoni@espia.it>
To: <ricvil@telesip.net>
Sent: Wednesday, January 07, 2004 8:57 AM
Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems

> Hi.
> Here's the ethereal dumps (running on the * server):
> one is when calling from the GS to a cisco phone,
> the other is viceversa.
>
> I dumped only the traffic between the * server and the GS
>
> Running firmware 1.4.30
>
> Matteo.
>
> Il mer, 2004-01-07 alle 14:44, TeleSIP ha scritto:
> > Hi Matteo,
> >
> > Send me the Ethereal SIP Trace and I will take a stab at it.
> >
> > Regards,
> > Andres.
> >
> > ----- Original Message ----- 
> > From: "Matteo Brancaleoni" <mbrancaleoni@espia.it>
> > To: <asterisk-users@lists.digium.com>
> > Sent: Wednesday, January 07, 2004 4:50 AM
> > Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
> >
> >
> > > Hi.
> > >
> > > I have the same issue with budgetones 102 (& 101) with
firmware
1.0.4.30> > > But happens also with .4.26 , .4.18 and .4.17 .
> > > Doing an ethereal trace, I noticed that the GS isn't
answering to OK's
> > > sent by asterisk when the ringed party answers (GS doesn't
not send
ACK> > > to the cpnnection confirmation), so after x seconds (6, more or
less)
> > > asterisk closes up the connection and so you get iCMP
unreacheable
> > > on the RTP port (that's ok since * closed the port).
> > >
> > > So GS must fix that... send ACK to the 200/OK of the connection
> > > confirmation.
> > >
> > > Pretty interesting is that when you call to another * channel
that's
not> > > SIP (like ZAP,CAPI) all works ok... only SIP<->SIP raises
the problem,
> > > or better only when the SIP call is initiated by the GS to
another SIP
> > > device. If the call is started from a cisco to the GS, all works
ok.
> > >
> > > Matteo.
> > >
> > > Il lun, 2004-01-05 alle 05:16, Mike Machado ha scritto:
> > > > I am trying to get the handytone 286 to make a very simple
call to *
and> > > > having problems. It registers with * just fine, but when I
place a
call> > > > (to echo test, for example), the RTP stream seems to have
problems
> > > > opening. Here is there error I get in *:
> > > >
> > > > WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt):
Maximum
retries> > > > exceeded on call
20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6
for> > > > seqno 0 (Response)
> > > >
> > > > When doing traces with ethereal, I see successful SIP and
SDP
> > > > handshakes, but when * sends handytone RTP packets, I see a
ICMP
Port> > > > Unreachable messages sent from Handytone to * regarding the
UDP RTP
> > > > packet. * then gives up and I see a BYE from *, which
handytone
acks.> > > >
> > > > Handytone config is default except obvious SIP registration
parameters.> > > > I also have a Sipura SPA2000 and everything works perfect
for that
one,> > > > same extension and everything (not at same time of course).
> > > >
> > > >
> > > > Both on same subnet, no NAT. I have two Handytones, both
exhibit
same> > > > symptoms.
> > > >
> > > > Anyone else have this problem?
> > > -- 
> > > Matteo Brancaleoni
> > > Espia System Administrator
> > > Email : mbrancaleoni@espia.it
> > > Web   : http://www.espia.it
> > > Phone : +39 02 70633354      - ext 201
> > > IAX(2): guest@213.140.14.155 - ext 201
> > > Iaxtel: 1-700-56-62458       - ext 201
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Matteo Brancaleoni
> Espia System Administrator
> Email : mbrancaleoni@espia.it
> Web   : http://www.espia.it
> Phone : +39 02 70633354      - ext 201
> IAX(2): guest@213.140.14.155 - ext 201
> Iaxtel: 1-700-56-62458       - ext 201
>-------------- next part --------------
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