atcomsystems.ca/forum Forums TDM Business Telephone Systems Samsung Telephones & Systems Randomly drops IP phone calls on Answer out SIP trunk

Help! I have a 7030 V4.60b with 3 SIP Trunk Lic, 3rd Party SIP Lic, 4 MGI Lic, remote IP Phone (SMT-i3105) and remote ATA (SPA122) over VPN for cordless phone & external ringer.

I place an outbound call from the SMT-i3105 (Ext211) to my office. I always hear the ringing. After the call is answered about 2 out of 5 times I get no 2-way voice. The other times the call rings, answers fine and 2-way voice.

I also get the same random problem on inbound calls to IP Phone (Ext.211). Problem also happens with Mobex calls off Ext.211 and also random on the ATA\cordless phone (Ext.217) which is paired to the Ext. 211.

I have full IP\Port on the router\firewall open to the ITSP. Tried with and without "fixup SIP" and "inspect SIP" options on router (Cisco881).

Customer is really stressing because half of their business calls are dropping and having to call customers back from CID log on cell phone.
Already contacted Samsung distributor support but no good solutions yet. Any ideas? Thanks in advance.

PS. I have the same setup on older 7030 with v.4.53 and different ITSP with no problems with IP phone or Mobex.

Codecs.. Try changing the codecs I would take a guess its codec neg. failing we are getting the SIP signally hence you get the ring coming from the Samsung however when they answer you are having issues Neg. the correct Codec, if you call inbound do you still get the same issues.

are the codec Neg. are normally at the receiver end.

Andma. In 5.2.13 I have "Codec Auto Nego = Enable", "SIP Trunking Codec PR1 = G.729" and "SIP Trunking Codec PR2 = G.711a", PR3 & PR4 set to Disable.

Same problem on inbound calls. First noticed it on Mobex\SIP off a IP Phone Ext. on inbound calls. Same thing on IP Phone & ATA. If I call my office vm from the IP Phone or ATA it will work fine with 2-way audio maybe 3 of 5 times. If I call the IP Phone or ATA from my cell phone it has the same results.

Is the Phone system in a DMZ off the router, here is a link how to program the DMZ on the 881

https://www.cisco.com/en/US/docs/security/pix/pix72/quick/guide/dmz_p.html

I've found that made a huge difference with out SIP Failing that if you send me a save I can have a quick look and compare it against a know worker.

but Sip is really Basic you have a Username, Password and Domain. if it registers generally its codec for firewall upsetting you

you can also try to Port forward UDP 16384 - 20384 direct to the MGI

Andma. No DMZ on the Cisco881 router. Just basic NAT and some ACLs for firewall. I have an ACL permitting the Full IP (all ports) from the ITSP. The SIP Trunk comes up fine on the 7030 and all the local iDCS28D phones work fine with calls in\out. I have a generic (GRE) site-to-site Cisco VPN connecting the 2nd site to a older Cisco 831 router. The 2nd site has the IP Phone & ATA and they are the only devices on the whole system having problems. I have exactly the same setup with 7030 & IP Phone over Cisco VPN working fine on V.4.53c and different ITSP.

The Codec Nego. sounds plausible. Is there a log on 7030 to show that the Codec Nego. failed? The Call Setup (SIP) is working because the call is ringing and connecting. Is the Codec Nego. again for the (RTP) phase? The ITP phone (2.7.1) is set to G.729a. I have noticed on the RTG Status (6.2.10) shows my inbound cell call to the ITP phone as G.711a. When watching calls going out of ITP the MPS Status (6.2.7) is showing codec as G.729a. It shows the same codec (G.729a) regardless if call connects correctly or if no 2-way audio.

I would like to do a wireshark capture on the Cisco811 WAN port to get more insight but not sure how to forward that traffic for remote wireshark monitor. Or, trying to capture ITP data on the remote router to 7030. I do this all the time on a Cisco managed switch with a SPAN port but not sure how it's done on these Cisco881 routers with built-in 4-port switch.

I think there is a way to use NetFlow on Cisco for a remote wireshark type monitor but I don't have a working example.

I might go and set everything (ITP & SIP Trunk) to G.711 only and see what happens. The Internet connections are cable with 50meg down & 2meg up with 3 SIP trunks so not worried about saving bandwidth with G.729a.

correction. My cell traffic inbound was G.711u. Also, SIP Trunking PR2=G.711u for North America.

Is it better to have the ITP setup for G.711u? It defaults to G.729a.

I'm not clear from your description. Are the calls dropping (like a hangup) on answer, or are you having one way speech issues? Hangup on answer is usually a codec issue. One way speech is usually an RTP routing/firewall issue.

Noisemarine. My mistake, the inbound\outbound calls to\from the remote ITP are connecting but with no 2-way audio about 50% of the time. My main location with the 7030 is on a 192.168.75.x network and the remote ITP is on a 192.168.76.x network. Routing is working between the subnets over a generic Cisco GRE VPN. I've added the remote subnet to MMC838. At the main location I have a ACL on the Cisco881 to allow the full IP (all ports) to passthru from ITSP so this should cover all the Ports needed for VoIP.

I'm not clear if after the Call Setup (SIP) on an inbound or outbound call to\from the ITP that (RTP) passes from the ITP to the 7030 over the VPN and then to ITSP or if the ITP passes (RTP) traffic directly out the local router to the Internet(not over VPN).

Both locations access the Internet directly, only site-to-site traffic is routed over the VPN. I'm not seeing any VoIP\NAT traffic on the remote router so I assume all ITP traffic goes directly to the 7030 over the VPN and then to ITSP. If I had router\firewall issues wouldn't the problem consistently not work? the 50\50 random occurrences is what is throwing me for a loop.

The Cisco881 has a embedded packet capture feature. So, I've setup an ACL to capture traffic based on ITSP public IP. I can then TFTP this capture buffer to my PC for Wireshark\SIP analysis. I'm going to try an capture a trace for successful and fail call to see if I can determine anything different on the traces.

I would like to test or rule out anything simple before I start trying to analyze traffic.

in 830 you can you confirm you have public W private and also confirm that your Public ip is entered.

I admit I'm still a little green on the Samsung way, but my past experience is call setup will be done via the PBX, but on connection RTP will route directly between the endpoints. I suspect that there might be an issue with the call setup routing out the internet connection from one site, and then suddenly appearing to come from the other once connected. I might be way off the mark, but it might be a simple test to add a route to the ITP end router that points traffic destined for the SIP provider across the VPN instead of directly out the local internet connection.


If you make a call and it fails, does it work on the second try? It might point at the firewall closing the ports after a period of time. The first attempt opens them, then they are available for the second attempt. I figure this could tie in with my musing above about the two internet connections being used at different parts of the call.


Good luck. This stuff has us tied in knots at times, too.