06-04-2004 08:59 AM - edited 03-13-2019 05:08 AM
Is it possible to have a gateway accept inbound voip dial-peer and send that call out another voip dial-peer? How might that configuration look?
Thanks!
06-10-2004 11:31 PM
Dear Prasanna,
Are you sure that calls are not even coming to AS?
In fact, maybe calls are coming trough PRI and then tries to dial telco again but telco is giving busy signal to those calls...
example:
1. someone dials PRI number 2720055 and he comes to the AS
2. num-exp or translation rule is applied and now AS dials telco number 160221,,,,1212121212
3. Telco gives busy signal
why?
1. number is really busy
2. number is invalid
3. pause time is not long enough
4. there are some user authentification which needs to be proccesed during a call
...
maybe you can try to debug and see if calls are coming to the AS and are they are going out of AS?
Regards,
Goran
06-11-2004 12:31 AM
Try this ...
translation-rule 1
rule 0 2720055 160221,,,,12121212
dial-peer voice 2171 pots
incoming called number 217232....
dial-peer voice 2172 pots
destination-pattern 2720055
translate-outgoing called 1
port 1/0/4:d
After you put this in the configuration, make a call and do 'show call hist voice brief' and see what dial-peers the call is matching, as Goran suggested.
K
06-11-2004 04:12 AM
Tried but still no movement
following obs I got :
sh call hist vo brie
dur hh:mm:ss tx:
IP
delay:
MODEMPASS
last
FR
sig:
ATM
sig:
Telephony
Proxy
anf>
bw:
tx:
tes>
rx:
es>
Total call-legs: 0
I think there is nothing in the above log
following is the dial-peer history :
sh dial-peer voice
VoiceEncapPeer2171
information type = voice,
tag = 2171, destination-pattern = `',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 2171, Admin state is up, Operation state is up,
incoming called-number = `217232....', connections/maximum = 0/unlimited
,
DTMF Relay = disabled,
huntstop = disabled,
in bound application associated: DEFAULT
out bound application associated:
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `',
forward-digits default
session-target = `', voice-port = `',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
VoiceEncapPeer2172
information type = voice,
tag = 2172, destination-pattern = `2720055',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 2172, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
huntstop = disabled,
in bound application associated: DEFAULT
out bound application associated:
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `',
forward-digits default
session-target = `', voice-port = `1/0/4:D',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Regds,
Prasanna
06-11-2004 05:18 AM
1. Were u able to successfully route calls through this gateway ? If yes How ?
2. Did u do PSTN -> IP on this gateway ?
3. R u able to do PSTN -> PSTN using different ports on this gateway ?
I can't see call coming in. Try to do 'debug isdn q931' and see if the call is coming in.
K
06-11-2004 09:15 PM
1.This AS 5800 is spcifically for Internet Dial up.
Uptill noe not used for any voice application.
2.No PSTN->IP on this gw
3.On all ports I am able to give ring to any telco no. by reverse telneting a modem from MICA modem pool
I tried to debug isdn q931.Output is :
sh debugging
ISDN:
ISDN Q931 packets debugging is on
ISDN Q931 packets debug DSLs. (On/Off/No DSL:1/0/-)
DSL 0 --> 31
1 1 1 1 1 1 - - - - - - - - - - - - - - - - - - - - - - - - - -
DSL 32 --> 55
- - - - - - - - - - - - - - - - - - - - - - - -
Following is the ISDN port status :
sh voice port 1/0/2:d
ISDN 1/0/2:D - 1/0/2:D
Type of VoicePort is ISDN
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Ringing Time Out is set to 180 s
Companding Type is A-law
Region Tone is set for US
Wait Release Time Out is 30 s
Station name None, Station number None
Voice card specific Info Follows:
DS0 channel specific status info:
IN OUT
PORT CH SIG-TYPE OPER STATUS STATUS TIP RING
For pstn->telco house->AS5800->pstn any voice card is required ?
Regds,
Prasanna
06-14-2004 12:20 AM
Kiran
Following is the debug output of isdn q931
107472: 9w5d: ISDN Se1/0/2:15: RX <- SETUP pd = 8 callref = 0x7097
107473: 9w5d: Sending Complete
107474: 9w5d: Bearer Capability i = 0x8090A3
107475: 9w5d: Channel ID i = 0xA98394
107476: 9w5d: Progress Ind i = 0x8283 - Origination address is non-ISDN
107477: 9w5d: Calling Party Number i = 0x2183, '2172721111', Plan:ISDN,
Type:National
107478: 9w5d: Called Party Number i = 0x81, '2055', Plan:ISDN, Type:Unkn
own
107479: 9w5d: ISDN Se1/0/2:15: TX -> RELEASE_COMP pd = 8 callref = 0xF097
107480: 9w5d: Cause i = 0x80AF - Quality of service unavailable
Regds,
Prasanna
06-07-2004 08:29 AM
Dan, you over-simplify things.
Every call going through a Cisco H.323 gateway is matched to both incoming and outgoing dial peers, even if the incoming dial peer is not explicitly defined. And Incoming-Called-Number is only 1 of several matching criteria that can be used.
A simply dial peer scheme can be built without this information. But a complex dial plan will send a person to the funny farm in short order unless these concepts are understood.
06-06-2004 10:19 PM
Voip - Voip is possible in 3745 ip-ip gateway running 12.3 or later. In other gateways you can only do ip-pstn or pstn-ip.
If you are using either AS5300 or AS5400 the best way to do, as explained by someother responder is to send the incoming ip to pstn and loop it back to another port and send it out as IP.
Another way of doing may be to write a TCL script, though I have not tried it myself, I think it can work.
Kiran.
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