[Serusers] problem with cisco 2600 to pstn
Greg Fausak
greg at august.net
Fri Apr 25 18:16:15 CEST 2003
Just FYI I though I'd post by BRI configuration
Here as well. It might help somebody.
I've got BRI ports running on one of my routers.
Here is my software version:
jabber#show ver
Cisco Internetwork Operating System Software
IOS (tm) 3600 Software (C3640-JS-M), Version 12.2(13b), RELEASE SOFTWARE
(fc1)
You have to have VWICs, not normla BRIs, my hardware includes:
FRU Part Number: NM-2V (this takes two VWIC cards)
TE BRI Voice daughter card (2 port, 4 channels) (VWIC 2B-ST)
Here is my configuration:
! Put the codecs you want to use here:
voice class codec 1
codec preference 1 g711ulaw
!
! Configure the BRI SPIDS like this:
interface BRI1/0
no ip address
isdn switch-type basic-ni
isdn spid1 97224277470101 2427747
isdn spid2 97224299170101 2429917
isdn incoming-voice voice
isdn sending-complete
!
voice-port 1/0/0
echo-cancel coverage 32
description 9722427747,9917
!
voice-port 1/0/1
echo-cancel coverage 32
description 9723234377,4378
!
! You have to configure outgoing patterns for each channel.
! These are US long distance, eg, 14085551212, preceeded by a 9
! I guess you could just use a pattern like T if you want everything
! To go out. The tricky part here is you have to recognise which
direction
! The call is going. You have inbound calls, and outbound calls.
! I use the '9' to queue to myself that the call came from somewhere
! Else and is destined for POTS. If you are using this for inbound
! You need to set up at least one 'direct-inward-dial' for each pots
! Port... I believe in this example below we allow inbound calls
! From just port 0, channel 0. I don't actually know what would happen
! With an inbound call on port 0, channel 1 in this configuration.
dial-peer voice 1 pots
destination-pattern 91..........
port 1/0/0
forward-digits 11
direct-inward-dial
!
dial-peer voice 2 pots
destination-pattern 91..........
port 1/0/1
forward-digits 11
!
dial-peer voice 1100 voip
destination-pattern T
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
max-forwards 15
retry invite 3
retry response 3
retry bye 6
retry cancel 3
timers trying 1000
sip-server ipv4:IP.OF.YOUR.SER.PROXY.SERVER
!
---greg
> -----Original Message-----
> From: serusers-admin at lists.iptel.org
> [mailto:serusers-admin at lists.iptel.org] On Behalf Of budi wibowo
> Sent: Friday, April 25, 2003 10:57 AM
> To: serusers at lists.iptel.org
> Subject: RE: [Serusers] problem with cisco 2600 to pstn
>
>
>
> codec set at the voip dial-peer with session target to
> sip server
>
> cmiiw
>
>
> budi
> --- Dan Austin <Dan_Austin at Phoenix.com> wrote:
> > On your pots dial-peer set the codec to g711alaw, or
> > configure your
> > SIP clients to use that codec.
> >
> > Dan
> >
> > -----Original Message-----
> > From: Yang Xiang
> > [mailto:yang.xiang at iitb.fraunhofer.de]
> > Sent: Friday, April 25, 2003 7:55 AM
> > To: serusers at lists.iptel.org
> > Subject: [Serusers] problem with cisco 2600 to pstn
> >
> >
> > Hi all,
> >
> > I am expericing problem with the cisco 2600, which
> > should function as
> > the sip2pstn gateway. If I try to complete a call
> > from a sip phone to
> > pstn, the router says:
> >
> --------------------------------------------------------------
> ----------
> > 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref
> > = 0x03
> > 00:15:49: Bearer Capability i = 0x8090A2
> >
> > ^^^^^^
> > 00:15:49: Channel ID i = 0x83
> > 00:15:49: Progress Ind i = 0x8181 - Call not
> > end-to-end ISDN,
> > may
> > have in-band info
> > 00:15:49: Calling Party Number i = 0x80,
> > 'yang2', Plan:Unknown,
> > Type:Unknown
> > 00:15:49: Called Party Number i = 0xC9,
> > '6091574', Plan:Private,
> > Type:Subscriber(local)
> > 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8
> > callref = 0x83
> > 00:15:49: Channel ID i = 0x89
> > 00:15:49: Progress Ind i = 0x8188 - In-band
> > info or appropriate
> > now
> > available
> > 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8
> > callref = 0x83
> > 00:15:49: Cause i = 0x80C1 - Bearer
> > capability not implemented
> >
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^
> > 00:15:49: Progress Ind i = 0x8188 - In-band
> > info or appropriate
> > now
> > available
> >
> --------------------------------------------------------------
> ----------
> > ----
> > ------------
> >
> > Please notice the line "Bearer Capability i =
> > 0x8090A2", the digits
> > "80" mean that this is a ITU voice call, "90" mean
> > circuit mode, 64 kbps
> > and "A2" is for G.711 u-law.
> >
> > So if I call the router from a normal telephone, the
> > debugging looks as
> > follows:
> >
> --------------------------------------------------------------
> ----------
> > ----
> > ----
> > 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref =
> > 0x01
> > 01:01:58: Bearer Capability i = 0x8090A3
> >
> > ^^^
> > 01:01:58: Channel ID i = 0x89
> > 01:01:58: Calling Party Number i = 0x0181,
> > '609157', Plan:ISDN,
> > Type:Unknown
> > 01:01:58: Called Party Number i = 0xC1,
> > '20', Plan:ISDN,
> > Type:Subscriber(local)
> > 01:01:58: High Layer Compat i = 0x9181
> > 01:01:58: ISDN BR1/0: Event: Received a VOICE call
> > from 609157 on B1 at
> > 64 Kb/s
> >
> --------------------------------------------------------------
> ----------
> > ----
> > ---------
> >
> > whereat the bearer capability is "0x8090A3". It
> > means that the
> > ISDN-switch of German Telekom uses G.711 a-law.
> >
> > I am afraid that is the reason why the sip-call
> > doesn't go through. But
> > I can't find any way to configure this.
> >
> > Has anybody in this mailinglist the same experience?
> >
> > Any hints would be very appreciated.
> >
> > Thanks
> >
> > yang
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
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