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New Member

PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Hi,can anyone help me with my VoIP setup? I currently have two branches, Branch-A and Branch-B in two different cities. I'm trying to setup an infrastructure whereby I can call into my local Branch via PSTN into my Cisco voice gateway... go through the Internet.. routed to the other Branch and out the Cisco voice gateway to the remote PSTN. Here,s what it looks like.

PSTNA -> (E1-PRI)CiscoGW -> Internet -> CiscoGW(T1-PRI) ->PSTNB

Branch-A has a CiscoGW with a NM-HDV(VWIC-1MFT-E1)connected to a T1 PRI and Branch-B CiscoGW has a NM-HDV(VWIC-1MFT-T1) connected to a T1 PRI. My problem is I can call from PSTNB thru to PSTNA and th person on PSTNA would ring and he would pickup and hear me from PSTNB.. but I don't hear anything from PSTNA. One way audio. I'm not using CM or anything so I don't think it's a routing issue. I tried pinging and pings went thru. I actually see the call being set up but no voice. I even used RTP cut-thru but nothing... anyone have any suggestions?

By the way, I'm just using simple Dial-Peer statements to point from one site to the other. Using Codec G.711. I actually followed steps from this link below but I made it even simplier as I don't have SIP configured.

Sorry for this question again but my last post got hijack by someone asking their own question and my question went unanswered.

Cisco Employee

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Make sure you bind your H323/SIP media and signalling packets to an interface you know can ping each other.

can you please post configs from both routers.

Then, make a call and do "show call active voice brief" on both routers. You will see the IP address of the RTP streams. Ping the other end router sourcing the correct ip address...

If your pings work, check if you have a firewall or a router in between that might be dropping RTP packets.

a "debug rtp packet" (caution verbose) will tell you if the router is sending packets out to the correct ip address and receiving packets.

New Member

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Hi, thanks for the suggestion I'm actually not using H323 or SIP. Here's my configs if you could please have a look and see if I'm missing anything.



Cisco Employee

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem


You are using H323.

You need to put in this command under the fastethernet interface:

h323-gateway v bind srcaddr

Also put

incoming called .

under the voip dial peers on both routers.

Then, make sure you can ping from one router to another.

Then, paste a "show call active voice brief" from both routers, when the call is in progress.

New Member

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Hi, thanks for your reply. I put in those commands and still 1 way audio. I have attached the whow commands. Can you please explain why I need H323? I'm only dialing peer to peer and no gateway is involved?



New Member

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Hi, can anyone help?... I'm banging my head against the wall here trying to figure out why I don't have 2 way voice!!!



Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Looks to me like a firewalling problem - either in terms of H323 proxy, NAT, or access rules preventing RTP from passing, Bug in IOS, Bug in Firewall.

We see Router1 sending to and receiving from IP Voip TX and RX & serial TX & RX are incrementing.

Total call-legs: 2

11DC : 100518816hs.1 +-1 pid:416 Answer 9055297186 connected

dur 00:00:00 tx:1626/260160 rx:161/25044

IP rtt:0ms pl:3120/160ms lost:0/1/0 delay:75/75/75ms g711ulw

11DC : 100518822hs.1 +-1 pid:2 Originate 0118613694852563 connecting

dur 00:00:00 tx:161/25044 rx:1626/260160

Tele 0/0:15 (14): tx:35240/32520/0ms g711ulaw noise:-84 acom:20 i/0:-79/-81 dm

Total call-legs: 2

But Router 2 is not recieving/relaying anything from Router1. (Note the voip RX field is 0/0, so the Serial TX field must also be 0/0.)

Total call-legs: 2

11DC : 43655466hs.1 +-1 pid:906 Answer 9055297186 connected

dur 00:00:00 tx:0/0 rx:163/25364

Tele 0/0:23:5: tx:37600/3220/0ms g711ulaw noise:-74 acom:20 i/0:-63/-81 dBm

11DC : 43656398hs.1 +-1 pid:1186 Originate 0118613694852563 connecting

dur 00:00:00 tx:163/25364 rx:0/0

IP x.x.x.x:16388 rtt:0ms pl:0/0ms lost:0/0/0 delay:75/75/75ms g711ulaw

Total call-legs: 2

All four legs are in a "connected" state, and you have not indicated any PBX/Callmanager or other systems in between, so the problem is not with control traffic connectivity. Further, Router2 is receiving control traffic, but not RTP. This seems to preclude routing as a problem. However, we can't be certain of that if there are H323 proxies in between.

Also, you mentioned this traffic crossing the internet. I'm assuming that you have a couple of firewalls up in between these gateways. And you may or may not have IPSEC running between them.

So, there are a few things that come to mind which could prevent Router2 from getting it's traffic:

1) Routing problem if there are H323 proxy(ies) in between these two gateways.

2) H323 Proxy functionality in the firewall, like PIX's "fixup h323 h225" is not correctly handling the H.245 portion of call setup. So Router2 ends up listening on the wrong UDP port.

3) This traffic is being NATted and NAT is altering the the H.245 portion of the setup so Router2 is listening on the wrong UDP port.

4) NAT is altering the UDP port of the RTP traffic so, even though Router2 is listening on the correct port, NAT has changed the UDP destination port to something else.

5) VoIP bug in either of the routers

6) VoIP or NAT bug in either of the firewalls

BTW, you mentioned that you are not using H323. Based on the configs you posted, you are indeed using H323.

HTH give you some leads for consideration.



New Member

Re: PSTN-A -->GW --> VoIP -->GW--> PSTN-B Problem

Hi, I followed your advice and found out it was a NAT problem!... Thanks so much!.. Now I have a new problem. I can dial from PSTN->RouterB->INTERNET-> RouterA->PSTN no problem.

But when I dial from the other side from

PSTN->RouterA I get dialtone and when I enter my destination digit string in I immediately get a fast busy. Does anyone have any ideas? I've attached my "show call active voice brief" and "show voice call sum" outputs.