Andrew Pogrebennyk wrote:
Hello Yusuf
yusuf wrote:
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco
So calls originate from the Asterisk side (registered users on SIP or
just ZAP phones), and they go out H323 or SIP to Cisco, where they go
out PRI.
I have the Asterisk side sorted :) (either H323 or SIP), I need help
in the Cisco side. Can anyone give me a brief HOW-TO or tutorial on
getting this (either SIP or H323) done on the Cisco side.
The link with sample Cisco config Hoah has sent is fine. It's well
commented etc, but... I do not recommend you to copy it entirely :)
> [...skipped...]
How do I specify that H323 or SIP must be for incoming calls, and
outgoing must go out on the E1.
Cisco is running IOS 12.1.5-12.2.13a
I realize this is alot of questions, so please bear with me :)
You seem to need a clear-cut explanation of dial-peer matching process
like http://www.cisco.com/warp/public/788/voip/in_dial_peer_match.html
or more complete guides:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/int_c/dpeer_c/dp_confg.htm
and
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fvvfax_c/vvfpeers.htm
I think I can help you deal with Cisco side once you have drafted a
clear setup.
Hi,
thanks for all the replies. We have got it mainly working, where we have Asterisk dial SIP to the
Cisco and Cisco goes E1 to the Telco. However, we can only make one call at a time, following calls
just hang. We have to reboot the Cisco to make another call :( . On the cisco, sc says this:
ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%>
<multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l>
dBm
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt:
<type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120
pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120
pkts>/<t120 bytes>
Total call-legs: 2
11DB : 30199hs.1 +-1 pid:0 Answer dj1 connecting
dur 00:00:00 tx:335/53441 rx:337/53920
IP 192.168.0.149:10612 rtt:0ms pl:3580/0ms lost:0/2/0 delay:64/64/65ms
g711ulaw
11DB : 30200hs.1 +-1 pid:1 Originate 0847889425 connecting
dur 00:00:00 tx:337/53920 rx:335/53441
Tele 0:0 (6): tx:6730/669/0ms g711ulaw noise:-60 acom:1 i/0:-58/-36
dBm
Is there something obvious we are missing?
--
thanks,
Yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
