Requirements for SIP Gateway - Page 2

clamasters · ‎03-28-2007

Hello all,

I'm pretty new to VoIP and have been given a little research to do. What do I need to configure on our router (Cisco 3845) to do the job that our AudioCodes device is doing (i.e. what do I need to configure to make the router a SIP/Voice gateway for Voice over Frame Relay connections)? We are using Interactive Intellegence software and Polycom phones for our phone system if that helps.

Arthur Kant · ‎04-04-2007

Hello,

I am with the same team/organization of the original poster and may be able to shed some additional light on our current setup and what we are trying to do.

Old Setup:

Hub and spoke setup with frame relay links to each site. Head end site has a 3845 each remote site has a 2811 connected via Frame-Relay and is configured for voice over frame with specific QoS rules. Our head end site additionally has a softswitch phone system which has polycom sip phones registered to it. Each remote site has an access code assigned to it so when you dial the access code only the end result is the ringing of one or more of the FXS ports on the remote router. The 3845 matches the access code sends to the remote site DLCI, remote router matches the same access code with POTS dial-peers and rings the line. The call is then delivered by each remote sites regular plain jane phone system (not supported by us). Any site can dial another sites access code and ring another remote site most often by getting access to the remote analog trunk on the router via dialing 9 or hitting a special key on their keysystem and then dialling another site's access code. The head end site with the softswitch used to participate in this also however it gained access to the 3845's dialplan by first traversing the AUDIO CODES SIP to Analog gateway. They would dial 88XX with the X's representing the remote sites access code. The softswitch would key on the 88 and send the call to the AUDIO CODES / which would then pick up a trunk on the T1 directly connected from the AUDIO CODES to the router

Present Time:

The AUDIO CODES device has failed at the head end site, and we would like to not replace it and have the softswitch be able to send and receive SIP calls with the 3845 and then "route" the calls based on our old dial plan via Voice Over Frame if possible. We have tested this by trying to dial site codes defined in the 3845's dial plan directly from a SIP phoned configured to use the 3845 as a proxy however 3845 is returning destination unknown. I can provide the SIP traces from these tests. Do we need to some how "accept" these SIP calls with an inbound dial-peer first? Can I have the 3845 take an SIP Invite and then deliver the call via already in place Voice over Frame Dial-Peers?

paolo bevilacqua · ‎04-04-2007

Hello,

As I indicated in my previous posts, you cannot call from a VoIP device to a VoFR directly - this is not supported by Cisco IOS.

The workaround is convert to TDM/POTS in between VoIP / VoFR. Your colleague was about to do that based on my previous configuration indications, and I understand he had already connected the required crossover cable between the two T1 ports.

Once you complete that configuration, is not much, you will be able to call from SIP softphone or softswitch to the remote sites.

clamasters · ‎04-05-2007

p.bevilacqua,

Thank you for all your help so far. I'm still looking into how to correclty configure the dial peers and a test phone to go through the PRI then get routed throught the IP network. Do you have any configuration examples that would help in this respect? Thank you.

Curtis

paolo bevilacqua · ‎04-05-2007

Hi,

After the two PRI interfaces are up, you only need three more dial-peers, one voip and two pots. The vofr DPs should be allright already.

dial-peer voice 10 voip

incoming called-number

destination-pattern

session-protocol sipv2

session-target ipv4:

dial-peer voice 20 pots

incoming called-number

destination-pattern

direct-inward-dial

port 0/1/1:23

dial-peer voice 30 pots

incoming called-number

destination-pattern

direct-inward-dial

port 0/1/0:23

clamasters · ‎04-12-2007

Ok, with your guidance a little bit of reading and some time on the phone with a tech, this is working. However, now our phone system lets us only make 1 concurent call at time over the FR interface. At this point I don't know if its a phone server problem or still a router configuration problem. What kind of show commands or debug commands can I do to view the current stat before, during and after a call is made. Thanks

Curtis

paolo bevilacqua · ‎04-12-2007

Glad to know things are starting working, I will love to see a resolved checkmark on this thred.

Now, the phone system doesn't know that you are using voFR or whatever, it just uses SIP, so it's trange no more than one calls goes through. Unless for some reason the harpinned PRI has only one channel working, we could look at "term mon" and "debug isdn q931" for this.

Then, for the call setup you have to look at "term mon" and "debug ccsip message" and this will tell how the calls is made and whay kind of error is returned. Also, try "debug voice dialpeer", but this one is a bit verbose and harder to interpret. To see the calls that are in place already, there is a bunch of commands, one is "show call active voice compact".

clamasters · ‎04-20-2007

Well, here is the debug output I'm getting from "debug isdn q931":

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:88210000@192.168.100.1:5060 SIP/2.0

To: <88210000>

From: "Turner_Associates" <>88@csystems.com>;tag=20112

Via: SIP/2.0/UDP 192.168.100.50:5061

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

CSeq: 1 INVITE

Contact: <88>

User-Agent: ININ-cssphone-87256560

Max-Forwards: 70

Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO

Accept: application/sdp

Accept-Encoding: identity

Content-Type: application/sdp

Content-Length: 205

v=0

o=ININ 649460000 649460000 IN IP4 192.168.100.50

s=Interaction

c=IN IP4 192.168.100.111

t=0 0

m=audio 2254 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.100.50:5061

From: "Turner_Associates" <>88@csystems.com>;tag=20112

To: <88210000>;tag=28838A2C-261C

Date: Fri, 20 Apr 2007 14:52:28 GMT

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.100.50:5061

From: "Turner_Associates" <>88@csystems.com>;tag=20112

To: <88210000>;tag=28838A2C-261C

Date: Fri, 20 Apr 2007 14:52:28 GMT

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

Apr 20 14:52:28.336: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:88210000@192.168.100.1:5060 SIP/2.0

To: <88210000>;tag=28838A2C-261C

From: "Turner_Associates" <>88@csystems.com>;tag=20112

Via: SIP/2.0/UDP 192.168.100.50:5061

CSeq: 1 ACK

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

User-Agent: ININ-cssphone-87256560

Max-Forwards: 70

Content-Length: 0

Basically I need to find out why I'm getting the "503 Service Unavailable" error and why it only happens sometimes. 4/5 calls I get this error. The rest of the time it goes through.

Curtis

paolo bevilacqua · ‎04-20-2007

Hi, this is "debug ccsip message". Can you also actually enable also "debug isdn q931" and take the call again ? This way we will be able to correlate the call flow.

Also please send output of "show controller t1" , "show isdn status", and "show dial-peer voice summary".