19 Apr
Asterisk and an AS5350 SIP peer
I’ve been playing with Asterisk lately, and needed to get out to the PSTN. Since I had a Cisco AS5350 with a PRI attached for incoming RAS, I figured I could use that. I also had some DIDs on the line that I could use to separate voice and data calls.
Googling around, I couldn’t find a good example of anyone who has done it, so I was on my own.
My goals here were simple:
- People dialing 555-1111 should be picked up by the AS5350’s internal modems
- People dialing 555-222X should be forwarded to my Asterisk box for handling
- The Asterisk box should be able to dial out using the PRI
Already the DIDs were assigned to this PRI, so that wasn’t a problem.
The configuration on the AS5350 is fairly straightforward once you get it working:
voice service voip sip ! Bind the SIP and RTP sessions to a known interface bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 ! Normal T1-PRI configuration using all channels controller T1 3/0 framing esf linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! Anything not picked up by a dial peer defaults to a modem call interface Serial3/0:23 isdn incoming-voice modem ! Use a data dial-peer to answer anything to 555-1111 with a modem dial-peer data 10 pots incoming called-number 5551111 ! Match inbound calls to the voice DIDs dial-peer voice 100 pots incoming called-number 555222. ! All digits are collected and sent along direct-inward-dial forward-digits extra ! The IP leg of the call… send to the sip server defined below dial-peer voice 200 voip destination-pattern 555222. session protocol sipv2 session target sip-server ! Any digits collected after the call has been answers get forwarded dtmf-relay sip-notify rtp-nte ! I want g711 ulaw for a codec codec g711ulaw ! Anything else gets sent out the PSTN ! Note you need the data dial-peer above to accept modem calls, otherwise ! this dial-peer will give dialtone to anyone who calls! dial-peer voice 1000 pots application session destination-pattern .T port 3/0:D forward-digits all ! define your SIP server sip-ua authentication username USERNAME password PASSWORD ! xxx.xxx.xxx.xxx represents the Asterisk box registrar ipv4:xxx.xxx.xxx.xxx expires 3600 sip-server ipv4:xxx.xxx.xxx.xxx
After that, it’s just a matter of configuring a sip peer in /etc/asterisk/sip.conf
I have a AS5300 used for RAS only.
Please, you could tell me if I need any Voice Feature Card in AS5300? or, maybe, special IOS version?
June 2nd, 2005 at 10:37 amI didn’t need any special hardware or software…
c5350-ik9s-mz.123-11.T.bin
2 FastEthernet interfaces
47 Serial interfaces
60 terminal lines
2 Channelized T1/PRI ports
Sean
June 2nd, 2005 at 10:39 amAsterisk and an AS5350 SIP peer
Asterisk and an AS5350…
June 12th, 2005 at 8:26 pmSean,
I found a mailing from you on regarding the uBR924. Did you ever get this working reliably? I think we have gone down a similar road with this sucker. First with MGCP and now I’m hung up on h323. I’m wondering if its my IOS (12.2.12l). Asterisk reports call setup H225 msg stating “connection refused” on the uBR, so the call clears. I will grab some captures to see what I can find, but I was hoping you could share some insight. — Kind Regards.
June 17th, 2005 at 6:41 pmYea, it works great.voice-rtr
>show ver
Cisco Internetwork Operating System SoftwareIOS ™ 920 Software (UBR920-K8O3V6Y5-M), Version 12.2(19c), RELEASE
SOFTWARE (fc2)
The image name is ubr920-k8o3v6y5-mz.122-19c.bin
Config below.
Do you get dialtone on the FXS side? Voice doesn’t work properly untilyou do the “cable-modem voip clock-internal” on the cable modem interface.
Also ISTR that you need a different load for MGCP andH.323…
Sean
interface Ethernet0
June 17th, 2005 at 8:29 pmip address 192.168.1.95 255.255.255.0
!
interface cable-modem0
shutdown
no cable-modem compliant bridge
cable-modem voip clock-internal
!
ip classless
no ip http server
no ip http cable-monitor
!
voice-port 0
input gain -2
output attenuation 0
!
voice-port 1
input gain -2
output attenuation 0
!
dial-peer voice 1 pots
destination-pattern 2001
port 0
!
dial-peer voice 100 voip
destination-pattern T
session target ipv4:192.168.1.10
dtmf-relay h245-alphanumeric
codec g711ulaw
!
dial-peer voice 2 pots
destination-pattern 2002
port 1
Sean,
The uBR is now working pretty solid; minus a bit of echo that I’ll troubleshoot another day. I ended up changing images. I’m now running ubr920-k1o3sv4y556i-mz.121-27a.bin. Its an older release, than the 12.2.19 release you’re running. It seems 12.2.19 has been deferred on CCO so I couldn’t download it.
I see now why you have specified the timer (T) character in your destination-pattern. It could just be this release, but the destination-pattern wildcard string syntax seems to differ from other IOS. I ended up creating, as you did, a default route with ‘destination-pattern .T’
I also added ‘timeouts interdigit 3′ to voice-port 0-1. This improved upon the 10sec delay upon dialing as a result of the timer string.
I probably blew 16 hours on this. I suppose its worth it considering that the uBR was free. Many thanks for your help.
June 20th, 2005 at 11:04 amplease note that if you set any DTMF-RELAY using ULAW/ALAW, you will not get any dtmf digit entered, because you need INBAND, and setting dtmf-relay put it in OUT OF BAND… greetings.
September 16th, 2005 at 1:51 pmSean,
Your posting was of great help, I hope you can answer few of my queries. I am using AS5350 for other hosted PBX solution, now I want to use the same gateway for Asterisk also. In sip-ua it points to my other server, is it necessary to specify asterisk in sip-ua.
Regards,
September 21st, 2005 at 12:54 pmManish Pramanik
I have a an as5350 that i wanted to install asterisk on for call loging or cdr.
MY as5350 is connected to 5 pri lines and i have multiple customers terminating calls from different ip addresses.
I am very new to this. Can someone please help?
September 30th, 2005 at 9:48 amHeya,
I suspect you’d need to get your customers to terminate on the asterisk
box which would make the calls out the AS5350. The way it is now, the
outbound dial-peers will get matched and never hit the asterisk box.
If all you want is CDR recording, look into “gw-accounting” using either
syslog or RADIUS.
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_command_reference_chapter09186a00800b3507.html
Sean
September 30th, 2005 at 9:54 amHi Sean,
Any chance you can publish the Asterisk side of this configuration. Specifically, the relevant sip.conf and extensions.conf
Thanks,
JPS
January 27th, 2006 at 3:21 amHi ill like to know if you can help me to get the oldest 12.0 IOS version for the ubr924 cablemodem? thanx
February 3rd, 2006 at 10:09 pmWe are a Cisco Partner and provide New and Factory Refurb AS5350, AS5350XM, AS5400, AS5400XM, AS5400HPX, AS5850 and much more. Visit us online or contact us today for a quote. We also have a Asterisk enigeer that is very good with the Cisco AS5XXX Gateways.
Shane Breen
February 8th, 2006 at 11:25 amDoretel Communications, Inc.
Director Of Sales & Marketing
Cisco Registered Partner
Office: 404.755.5721
Fax: 404.521.4639
sbreen@doretel.com
AIM: shanebreen2003
www.doretel.com
I have a problem with dtmf coming from Asterics to AS5300.
dtmf-relay rtp-nte is my setting at the Cisco side.
Sometimes it works, sometimes it not.
Sometimes Cisco recognise DTMF if I dial slow.
“Sometimes” means that during 15 min testing
it could work and not work many times.
dial-peer voice 107718 voip
incoming called-number 107……….
voice-class codec 999
session protocol sipv2
session transport udp
dtmf-relay rtp-nte
playout-delay mode adapt
playout-delay nominal 120
playout-delay maximum 1000
playout-delay minimum high
playout-delay fax 700
fax rate 7200
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
Please help,
February 8th, 2006 at 12:00 pmMichael
hello,
anybody could give me a hand ??. I want to configure a Cisco AS5350 as voice gateway for a IP Telephony solution. A PRI is connected to the AS5350 and in the LAN I have IP Phones and Asterisk. Where do I can find info on how to make the whole thing work ???
Thanks
March 3rd, 2006 at 12:29 pmlow_eve
low_eve. Give me a call (I know it’s bad but I don’t check my email any more) and I’ll give you a step by step on how to get your asterisk/Cisco working. I’ve just put together a project like that so I can save you tons of time (705)878-0997
March 11th, 2006 at 11:23 pmAnyone have a template on how to get a cisco 2431 running sip to register with Asterisk? I have it working fine using another sip server, but can not get the cisco to register with asterisk. Basically the cisco will be the gateway to the PSTN.
March 22nd, 2006 at 6:18 pmConfigure in the Cisco site:
voice-port 2/0
April 3rd, 2006 at 2:25 pmvad
cptone VE
timing percentbreak 60
!
dial-peer voice 15 voip
destination-pattern 2803
session protocol sipv2
session target ipv4:XXX.XXX.XXX.XXX
!
sip-ua
sip-server ipv4:XXX.XXX.XXX.XXX
Is there anyway to use the AS5350 as a telephony bridge to use in conferencing instead of pushing the calls out of the other side (i.e. put the calls into a container on the server as one conference call)? Any help would be appreciated.
June 6th, 2006 at 9:31 amI have AS5350. I configured it to work with faxes using T.37. It receives faxes from PSTN and send as an email attachment. But it refuses to send faxes from e-mail TIFF attachments to PSTN. But if I take an attachment I received from AS5350 and send it in e-mail – fax transmission to PSTN is O.K. Please, give me advice – what to do to make faxes work in both directions ?
July 14th, 2006 at 1:07 amHi Sean!
You configured the cisco as5350 for outbound sip calls. What i want is to accept sip calls on as5350 and route them to a PRI (ISDN PBX).
Any advise please.
Please leave me a number where i can reach you.
August 22nd, 2006 at 9:53 amHi,
I am using a Cisco AS5350 as VOIP gateway and I have a problem. 80% of incoming calls come up with the number of head, and not with the telephone number assigned to each customer in our Asterisk. Where can be the problem? I am sure that problem is configuration AS, but not it can be.
Thanks.
AS configuration is as follows:
Current configuration : 10675 bytes
January 28th, 2008 at 3:55 am!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
boot-start-marker
boot system flash c5350-is-mz.123-11.T9.bin
no boot startup-test
boot-end-marker
!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
no aaa new-model
ip subnet-zero
!
!
no ip cef
!
!
isdn switch-type primary-net5
!
!
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
modem passthrough nse codec g711alaw
sip
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw
!
!
!
!
controller E1 3/0
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 3/1
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 3/2
!
controller E1 3/3
!
controller E1 3/4
!
controller E1 3/5
!
controller E1 3/6
!
controller E1 3/7
!
!
interface FastEthernet0/0
ip address 192.168.100.253 255.255.255.0 secondary
ip address 192.168.100.2 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0
no ip address
shutdown
clockrate 2000000
!
interface Serial3/0
no ip address
shutdown
!
interface Serial0/1
no ip address
shutdown
clockrate 2000000
!
interface Serial3/0:15
no ip address
isdn switch-type primary-net5
isdn incoming-voice modem
no cdp enable
!
interface Serial3/1:15
no ip address
isdn switch-type primary-net5
isdn incoming-voice modem
no cdp enable
!
interface Async1/00
no ip address
!
interface Async1/01
no ip address
!
(…)
!
interface Async2/106
no ip address
!
interface Async2/107
no ip address
!
interface Group-Async0
no ip address
no group-range
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.100.3
no ip http server
!
!
!
control-plane
!
!
!
voice-port 3/0:D
!
voice-port 3/1:D
!
!
!
dial-peer voice 200 voip
destination-pattern .
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1000 pots
application session
destination-pattern .
translate-outgoing called 1
direct-inward-dial
port 3/0:D
forward-digits all
!
dial-peer voice 1001 pots
application session
destination-pattern .
translate-outgoing called 1
direct-inward-dial
port 3/1:D
forward-digits all
!
!
sip-ua
sip-server ipv4:10.10.0.54
!
ss7 mtp2-variant Bellcore 0
ss7 mtp2-variant Bellcore 1
ss7 mtp2-variant Bellcore 2
ss7 mtp2-variant Bellcore 3
!
line con 0
line aux 0
line vty 0 4
login local
line 1/00 2/107
modem InOut
!
scheduler allocate 10000 400
ntp clock-period 17179943
ntp server 202.182.192.94
end
This post is very similar to what I am trying to accomplish with an AS5300. THe problem I have though is that inbound calls from the PSTN when trying to be sent to sip-server they are rejected with SIP/2.0 488 Not acceptable here
Aterisk 1.2.12.1
IOS ™ 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE SOFTWARE (fc5)
I know this thread is old but has anyone see this issue ? Outbound calls from PBX to cisco work fine.
October 9th, 2008 at 1:11 am