Adam Rothschild
2005-Mar-19 15:48 UTC
[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform
Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk understands. Caller ID _number_ works fine. (I'm guessing this has something to do with the 'remote-party-id' header, but I've tried with it both enabled and disabled in the 'sip-ua' IOS configuration stanza.) 2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk -> Dial([...],,r) calls placed. Music on hold ([...],,m) works fine. Clues appreciated, on or off list. Relevant 'show' output and configuration snippets below... Thanks, and my apologies for the cross-posting, -a --==-- Router#sh ver Cisco Internetwork Operating System Software IOS (tm) 5350 Software (C5350-IK9S-M), Version 12.3(13), RELEASE SOFTWARE (fc2) Router#show spe ver IOS-Bundled Default Firmware-Filename Version Firmware-Type ===================================== ============ ============system:/ucode/spe_firmware-1 0.10.2.2 SPE firmware On-Flash Firmware-Filename Version Firmware-Type ===================================== ============ ============ SPE-# Type Port-Range Version UPG Firmware-Filename 1/00 CSMV6 0000-0005 0.10.2.2 N/A ios-bundled default [...] Router#show conf [...] spe country t1-default isdn switch-type primary-ni ! voice hunt user-busy voice call send-alert voice call convert-discpi-to-prog voice rtp send-recv voice service voip fax protocol pass-through g711alaw h323 sip bind all source-interface FastEthernet0/0 controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 interface Serial3/0:23 description T1 to CLEC no ip address load-interval 30 isdn switch-type primary-ni isdn incoming-voice modem no cdp enable voice-port 3/0:D bearer-cap Speech dial-peer voice 1 pots incoming called-number 21255512[00-50] direct-inward-dial ! dial-peer voice 100 voip destination-pattern 21255512[00-50] progress_ind setup enable 3 session protocol sipv2 session target ipv4:10.10.10.10 codec g711ulaw no vad ! dial-peer voice 1000 pots destination-pattern .......... port 3/0:D sip-ua retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server ipv4:10.10.10.10 line 1/00 1/59 no flush-at-activation no modem InOut transport input all line 2/00 2/59 no flush-at-activation no modem InOut transport input all
Kevin P. Fleming
2005-Mar-19 17:08 UTC
[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform
Adam Rothschild wrote:> 1) Caller ID name data comes in on the PRI, but doesn't appear to get > handed off to the Asterisk server via SIP, at least not in any > format that Asterisk understands. Caller ID _number_ works fine. > > (I'm guessing this has something to do with the 'remote-party-id' > header, but I've tried with it both enabled and disabled in the > 'sip-ua' IOS configuration stanza.)Turn on RPID generation in your AS5300, and then set "trustrpid=yes" in the SIP user entry for the gateway in Asterisk.> 2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk > -> Dial([...],,r) calls placed. Music on hold ([...],,m) works > fine.Play with the 'progressinband' setting in Asterisk to see if you can affect this; it has three settings 'yes', 'no', and 'never'. It's likely that what is happening is that Asterisk is sending '180 Ringing' and the AS5300 is not generating ringback itself or asking the switch on the other end of the PRI to do it (or farther up the PSTN chain).
Oswaldo Arratia
2005-Mar-19 20:43 UTC
[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access ServerPlatform
Did you solve your problem? I have the same setup and it works for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam Rothschild Sent: Saturday, March 19, 2005 5:49 PM To: asterisk-users@lists.digium.com Cc: asr@latency.net Subject: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access ServerPlatform Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk understands. Caller ID _number_ works fine. (I'm guessing this has something to do with the 'remote-party-id' header, but I've tried with it both enabled and disabled in the 'sip-ua' IOS configuration stanza.) 2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk -> Dial([...],,r) calls placed. Music on hold ([...],,m) works fine. Clues appreciated, on or off list. Relevant 'show' output and configuration snippets below... Thanks, and my apologies for the cross-posting, -a --==-- Router#sh ver Cisco Internetwork Operating System Software IOS (tm) 5350 Software (C5350-IK9S-M), Version 12.3(13), RELEASE SOFTWARE (fc2) Router#show spe ver IOS-Bundled Default Firmware-Filename Version Firmware-Type ===================================== ============ ============system:/ucode/spe_firmware-1 0.10.2.2 SPE firmware On-Flash Firmware-Filename Version Firmware-Type ===================================== ============ ============ SPE-# Type Port-Range Version UPG Firmware-Filename 1/00 CSMV6 0000-0005 0.10.2.2 N/A ios-bundled default [...] Router#show conf [...] spe country t1-default isdn switch-type primary-ni ! voice hunt user-busy voice call send-alert voice call convert-discpi-to-prog voice rtp send-recv voice service voip fax protocol pass-through g711alaw h323 sip bind all source-interface FastEthernet0/0 controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 interface Serial3/0:23 description T1 to CLEC no ip address load-interval 30 isdn switch-type primary-ni isdn incoming-voice modem no cdp enable voice-port 3/0:D bearer-cap Speech dial-peer voice 1 pots incoming called-number 21255512[00-50] direct-inward-dial ! dial-peer voice 100 voip destination-pattern 21255512[00-50] progress_ind setup enable 3 session protocol sipv2 session target ipv4:10.10.10.10 codec g711ulaw no vad ! dial-peer voice 1000 pots destination-pattern .......... port 3/0:D sip-ua retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server ipv4:10.10.10.10 line 1/00 1/59 no flush-at-activation no modem InOut transport input all line 2/00 2/59 no flush-at-activation no modem InOut transport input all _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Oswaldo Arratia
2005-Mar-20 09:45 UTC
[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access ServerPlatform
Here is what should work for you.
In your Cisco
dial-peer voice xxxxx voip
huntstop
destination-pattern xxxxx <- Extension number you want to dial
progress_ind setup enable 3
session protocol sipv2
session target ipv4:y.y.y.y <- Your * IP
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
In your *:
EXTENSIONS.CONF:
exten => _xxxxx,1,Dial(SIP/${EXTEN},45,f)
[GW1]
include => extensions
SIP.CONF:
[general]
;progressinband=no
[GW1]
amaflags=billing
type=peer
host=x.x.x.x <- Ip of your cisco gateway
defaultip=x.x.x.x <- Ip of your cisco gateway
port=5060
context=GW1
disallow=all
allow=ulaw
allow=g729
That should solve your ringback problem
Of course, you have to adjust codecs, ports and context names to match your
setup.
Hope this helps.
Oswaldo A.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam
Rothschild
Sent: Saturday, March 19, 2005 5:49 PM
To: asterisk-users@lists.digium.com
Cc: asr@latency.net
Subject: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access
ServerPlatform
Hello,
I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is
talking to an Asterisk server via SIP for call origination and termination.
Seems simple enough, and it works for the most part,
but:
1) Caller ID name data comes in on the PRI, but doesn't appear to get
handed off to the Asterisk server via SIP, at least not in any
format that Asterisk understands. Caller ID _number_ works fine.
(I'm guessing this has something to do with the 'remote-party-id'
header, but I've tried with it both enabled and disabled in the
'sip-ua' IOS configuration stanza.)
2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk
-> Dial([...],,r) calls placed. Music on hold ([...],,m) works
fine.
Clues appreciated, on or off list. Relevant 'show' output and
configuration
snippets below...
Thanks, and my apologies for the cross-posting, -a
--==--
Router#sh ver
Cisco Internetwork Operating System Software IOS (tm) 5350 Software
(C5350-IK9S-M), Version 12.3(13), RELEASE SOFTWARE (fc2)
Router#show spe ver
IOS-Bundled Default Firmware-Filename Version Firmware-Type
===================================== ============
============system:/ucode/spe_firmware-1 0.10.2.2 SPE
firmware
On-Flash Firmware-Filename Version Firmware-Type
===================================== ============ ============
SPE-# Type Port-Range Version UPG Firmware-Filename
1/00 CSMV6 0000-0005 0.10.2.2 N/A ios-bundled default
[...]
Router#show conf
[...]
spe country t1-default
isdn switch-type primary-ni
!
voice hunt user-busy
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
fax protocol pass-through g711alaw
h323
sip
bind all source-interface FastEthernet0/0 controller T1 3/0 framing esf
linecode b8zs pri-group timeslots 1-24 interface Serial3/0:23 description
T1 to CLEC no ip address load-interval 30 isdn switch-type primary-ni
isdn incoming-voice modem no cdp enable voice-port 3/0:D bearer-cap Speech
dial-peer voice 1 pots incoming called-number 21255512[00-50]
direct-inward-dial !
dial-peer voice 100 voip
destination-pattern 21255512[00-50]
progress_ind setup enable 3
session protocol sipv2
session target ipv4:10.10.10.10
codec g711ulaw
no vad
!
dial-peer voice 1000 pots
destination-pattern ..........
port 3/0:D
sip-ua
retry invite 4
retry response 3
retry bye 2
retry cancel 2
sip-server ipv4:10.10.10.10
line 1/00 1/59
no flush-at-activation
no modem InOut
transport input all
line 2/00 2/59
no flush-at-activation
no modem InOut
transport input all
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