[Asterisk-Users] Cisco AS5XXX to asterisk debugging.

Jorge Verastegui G jorge at redcetus.com
Sun Dec 12 17:23:09 MST 2004 

Hi
thanks for your help .


I do not have direct access to the Cisco, but I believe that he is
AS5300

The ios version is 12.2

and the cisco dum config is:

GWSCZ01>en
Password:
GWSCZ01#sh run
Building configuration...

Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58 UTC Mon Apr 16 2001
! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname GWSCZ01
!
no boot startup-test
logging queue-limit 100

!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
ip subnet-zero
ip cef
no ip domain lookup
!
isdn switch-type primary-net5
!
!
voice service voip
 fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
 sip
!
voice class codec 11
 codec preference 1 g729br8
 codec preference 2 g729r8
 codec preference 3 gsmfr
 codec preference 4 g726r32
 codec preference 6 g726r16
 codec preference 7 g723r63
 codec preference 8 g723r53
 codec preference 9 g726r24
 codec preference 10 g723ar63
 codec preference 11 g723ar53
 codec preference 12 g711ulaw
 codec preference 13 g711alaw
 codec preference 14 clear-channel
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
voice source-group cisco
 access-list 8
 carrier-id target cisco
!
!
!
fax interface-type fax-mail
mta receive maximum-recipients 0
!
!
!
controller E1 7/0
 framing NO-CRC4
 line-termination 75-ohm
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
 cas-custom 0
  country bolivia
!
controller E1 7/1
 line-termination 75-ohm
 pri-group timeslots 1-31
!
controller E1 7/2
 line-termination 75-ohm
 pri-group timeslots 1-31
 description Embratel
 --More--
!
!
interface FastEthernet0/0
 ip address y.y.y.y 255.255.255.224
 duplex auto
 speed auto
 no cdp enable
 h323-gateway voip interface
 h323-gateway voip id GK01 ipaddr y.y.y.z 1719
 h323-gateway voip h323-id GWSCZ01
 h323-gateway voip tech-prefix 2032#
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 y.y.y.v
no ip http server
!
!
!
!
!
!
call rsvp-sync
!
voice-port 7/0:0
 compand-type a-law
!
voice-port 7/1:D
!
voice-port 7/2:D
!
voice-port 7/3:0
 compand-type a-law
!
voice-port 7/4:0
 compand-type a-law
!
voice-port 7/5:0
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 4444 voip
 destination-pattern 44T
 voice-class codec 11
 session protocol sipv2
 session target sip-server
 session transport udp
!
dial-peer voice 7777 pots
 destination-pattern 7777T
 direct-inward-dial
 port 7/0:0
!
sip-ua
 retry invite 3
 retry cancel 2
 sip-server ipv4:x.x.x.x
!


On Sun, 2004-12-12 at 20:07, Hatzis, Michael wrote:
> What's the cisco box,52 / 53; version ios? can you post a config dump?
> 
> Regards
> 
>  
> 
> Michael Hatzis
> 
>  0421 476 211
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jorge
> Verastegui G
> Sent: Monday, 13 December 2004 10:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> 
> Excuse the insistence but I am more than one week with this problem, and
> I do not have any idea to solve it.
> 
> You know if the configuration with GK in the Cisco, can be interfering
> with the RTP traffic?  
> 
> 
> Thanks in advance
> 
> 
> 
> On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
> > Pls, post your Cisco and * config files.
> > 
> >  
> > 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jorge
> > Verastegui G
> > Sent: Friday, December 10, 2004 12:30 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
> > 
> > Hi, 
> > 
> > I have a serious problem to configure Cisco AS5XXX and Asterisk , 
> > 
> > I trying to use asterisk for 
> > 
> > PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) 
> > 
> > (No Nat, no Firewall)
> > 
> > I hear (on the PSTN(A)) clearly what the other person is saying, but
> the
> > other person (on the PSTN(B) side) hears nothing from PSTN(A).
> > 
> > I use tcpdump for debug de rtp trafic, and ouput contains 
> > 
> > 
> > 19:06:00.741293 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.763133 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.740415 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.810312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.860314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:00.980351 IP (tos 0x0, ttl  64, id 180, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.000313 IP (tos 0x0, ttl  64, id 181, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none],
> > proto 17, length: 164) y.y.y.y.18975 > x.x.x.x.19927: UDP, length 136
> > 19:06:01.020312 IP (tos 0x0, ttl  64, id 182, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.040302 IP (tos 0x0, ttl  64, id 183, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.060343 IP (tos 0x0, ttl  64, id 184, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.083311 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.128314 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.130316 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.165318 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 19:06:01.186312 IP (tos 0x0, ttl  64, id 179, offset 0, flags [DF],
> > proto 17, length: 60) x.x.x.x.19926 > y.y.y.y.18974: [no cksum] UDP,
> > length 32
> > 
> > 
> > Where
> > 
> >  x.x.x.x = ip address of Astersik
> >  y.y.y.y = ip address of Cisco
> > 
> > 
> > Two types of codecs were proven ( ulow, g729 ).
> > 
> > When use the Asterisk with Sip phones everything works well.
> >  
> >         SipPhone------>Asterisk------->PSTN(B)     
> > 
> > The configurations, are the usual ones (from the wiki). the version of
> > asterisk is 1.0.3, the linux is  FC2.
> > 
> > 
> > Please help me.  
> 
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