IP 706 not registering across Cisco ASA

[hubtech]

Well, The phone has now downloaded all of its config from the ftp server, thanks for the help!!!!

Now i'm having an issue getting the sip traffic to flow.  The phone will not register.  This is the result of Debug Sip on the Cisco firewall.  Has anyone come across this?

BH-Picayune(config)# sipSIP::REGISTER received from inside:192.168.40.20/5060 to outside:Adtran/5060

SIP::regex engine has reached end of packet

SIP::Found CSeq 514253258 REGISTER

SIP::Found URI in request line "sip:x.x.x.x-sip:5060" (21)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found From addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found From addr tag "f7a6dbdd73a8363" (15)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found To addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found Via branch "z9hG4bKaa819bde0" (16)

SIP::Found Via addr "SIP/2.0/UDP 192.168.40.20:5060;branch=z9hG4bKaa819bde0" (54)

SIP::Found Max-Forwards 70

SIP::Found Call-ID a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

SIP::Found valid SIP URI: sip:402@192.168.40.20:5060

SIP::Found Contact sip:402@192.168.40.20:5060

SIP::Found Content-length 0

Found port 5060

Found port 5060

Via Port 5060

Found port 5060

Found port 5060

Found port 5060

SIP::Found Expires, 3600 seconds

SIP::Found User-Agent

Created SIP Transaction for inside:192.168.40.20/5060 to outside:Adtran/5060

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

SIP::Updating xlate timeout for 192.168.40.20/5060 to 1:00:00

SIP:: Forward 527 bytes, total 527

BH-Picayune(config)# sipSIP::REGISTER received from inside:192.168.40.20/5060 to outside:Adtran/5060

SIP::regex engine has reached end of packet

SIP::Found CSeq 514253258 REGISTER

SIP::Found URI in request line "sip:x.x.x.x-sip:5060" (21)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found From addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found From addr tag "f7a6dbdd73a8363" (15)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found To addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found Via branch "z9hG4bKaa819bde0" (16)

SIP::Found Via addr "SIP/2.0/UDP 192.168.40.20:5060;branch=z9hG4bKaa819bde0" (54)

SIP::Found Max-Forwards 70

SIP::Found Call-ID a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

SIP::Found valid SIP URI: sip:402@192.168.40.20:5060

SIP::Found Contact sip:402@192.168.40.20:5060

SIP::Found Content-length 0

Found port 5060

Found port 5060

Via Port 5060

Found port 5060

Found port 5060

Found port 5060

SIP::Found Expires, 3600 seconds

SIP::Found User-Agent

SIP::State Machine: New Request '514253258 REGISTER' received on existing transaction, Deleting existing transaction

Deleted SIP Transaction

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

Created SIP Transaction for inside:192.168.40.20/5060 to outside:Adtran/5060

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

SIP::Updating xlate timeout for 192.168.40.20/5060 to 1:00:00

SIP:: Forward 527 bytes, total 527

SIP::REGISTER received from inside:192.168.40.20/5060 to outside:Adtran/5060

SIP::regex engine has reached end of packet

SIP::Found CSeq 514253258 REGISTER

SIP::Found URI in request line "sip:x.x.x.x-sip:5060" (21)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found From addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found From addr tag "f7a6dbdd73a8363" (15)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found To addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found Via branch "z9hG4bKaa819bde0" (16)

SIP::Found Via addr "SIP/2.0/UDP 192.168.40.20:5060;branch=z9hG4bKaa819bde0" (54)

SIP::Found Max-Forwards 70

SIP::Found Call-ID a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

SIP::Found valid SIP URI: sip:402@192.168.40.20:5060

SIP::Found Contact sip:402@192.168.40.20:5060

SIP::Found Content-length 0

Found port 5060

Found port 5060

Via Port 5060

Found port 5060

Found port 5060

Found port 5060

SIP::Found Expires, 3600 seconds

SIP::Found User-Agent

SIP::State Machine: New Request '514253258 REGISTER' received on existing transaction, Deleting existing transaction

Deleted SIP Transaction

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

Created SIP Transaction for inside:192.168.40.20/5060 to outside:Adtran/5060

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

SIP::Updating xlate timeout for 192.168.40.20/5060 to 1:00:00

SIP:: Forward 527 bytes, total 527

SIP::REGISTER received from inside:192.168.40.20/5060 to outside:Adtran/5060

SIP::regex engine has reached end of packet

SIP::Found CSeq 514253258 REGISTER

SIP::Found URI in request line "sip:x.x.x.x-sip:5060" (21)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found From addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found From addr tag "f7a6dbdd73a8363" (15)

SIP::Found valid SIP URI: sip:402@x.x.x.x-sip:5060

SIP::Found To addr "sip:402@x.x.x.x-sip:5060" (25)

SIP::Found Via branch "z9hG4bKaa819bde0" (16)

SIP::Found Via addr "SIP/2.0/UDP 192.168.40.20:5060;branch=z9hG4bKaa819bde0" (54)

SIP::Found Max-Forwards 70

SIP::Found Call-ID a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

SIP::Found valid SIP URI: sip:402@192.168.40.20:5060

SIP::Found Contact sip:402@192.168.40.20:5060

SIP::Found Content-length 0

Found port 5060

Found port 5060

Via Port 5060

Found port 5060

Found port 5060

Found port 5060

SIP::Found Expires, 3600 seconds

SIP::Found User-Agent

SIP::State Machine: New Request '514253258 REGISTER' received on existing transaction, Deleting existing transaction

Deleted SIP Transaction

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

Created SIP Transaction for inside:192.168.40.20/5060 to outside:Adtran/5060

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253258 REGISTER

Branch: z9hG4bKaa819bde0

SIP::Updating xlate timeout for 192.168.40.20/5060 to 1:00:00

SIP:: Forward 527 bytes, total 527

Deleted SIP Transaction

Call-ID: a7de06334dba821584ebca40b65702b9@192.168.40.20 (46)

CSeq: 514253253 REGISTER

Branch: z9hG4bK01ab1b407

SIP::Timeout, deleting transaction

[Anonymous]

Since this is another topic I branched this to a new thread.

If you run a debug sip stack messages register on the NetVanta 7100 do you ever see it attempt to register?  If not I would suggest looking at the firewall and SIP config on the Cisco as well as the firewall configuration of the 7100 if enabled.  I would assume the VPN/firewall configuration of the 7100 is ok since the phone was able to download files. 

Thanks,

Matt

[hubtech]

I was able to get the phone to register. Access list on the netvanta. Now I get 5 whole seconds of talk time.....then it goes silent on both sides but the call stays connected.

What a wild ride lol

Sent from my iPhone

[Anonymous]

It sounds like it could be another firewall related issue if the RTP works initially and then goes away while the call appears to be up on both ends.  I am not sure it will help us but it may be worth getting a debug sip stack messages and debug voice verbose for a test call where this issue is recreated.  If you can collect that along with a current copy of the configuration in a .zip file and put it on our FTP server with the instructions below I would be happy to review it.

Open Internet Explorer web browser on their PC
Type the following URL:  ftp://ftp.adtran.com

Press the Alt key, click View, and then click Open FTP Site in Windows Explorer

Double-click the "Incoming" folder
Drag and drop files from PC into the Internet Explorer window

Thanks,

Matt

[nogie]

Hello Matt, I am having a similar problem and would like to know how you solved this.. Please assist. Thanks

Nogie

[Anonymous]

Nogie,

I am not sure that this was resolved.  It sounded like it was a firewall issue from the description.  For your case do you get audio between the two phones initially and then it cuts out?  If you can provide more details on the steps you take to reproduce the issue as well as the symptoms that would help so I can make a recommendation.

Thanks,

Matt

[nogie]

Matt,

Thanks for the response.. I get audio between the sip servers / gateways for approx 15-25 sec and it just drops off. I have a trace from the session if you want to see it. What beats me here is that calls to another interconnected server using the same softphone from my side, functions normally. I can send it via email if you want.

Thanks,

Nogie

[hubtech]

i've zipped and uploaded the asa and adtran's running configs.

[hubtech]

full adtran config sent now. sorry about thie first one.

[Anonymous]

Nogie,

I would need to see the current configuration of the unit and the output of a debug voice verbose as well as a debug sip stack messages (both enabled at the same time within the same debug).  If you have an ADTRAN AOS Voice product at both sides I would like to see the debug and config from both sides at the same time when the issue is recreated.  You can combine everything in a .zip file and place it on the FTP server with the instructions listed earlier in this thread.

Thanks,

Matt

[Anonymous]

hubtech,

I am going to need to see the debugs requested earlier to be able to assist further.  Without seeing a complete debug the only thing I can recommend offhand is a firmware upgrade since you are currently on A4.03.  If you upgrade the chassis firmware, make sure to read the release notes to determine the required phone firmware that should accompany the NetVanta 7100 firmware.

Thanks,

Matt

[hubtech]

OK, i've uploaded everything.

[Anonymous]

hubtech,

Previously you were using extension 402, but I did not see that in this debug.  It seems that debug sip stack level debug was enabled instead of debug sip stack messages.  Can you get another debug with both debug sip stack messages and debug voice verbose enabled at the same time in the same terminal session?  You will need to make sure your terminal program has a lot of scroll-back buffer to catch everything without clipping off part of the debug.  When you resubmit the files to the FTP server, please include a call flow description.  Ex: extension 402 dials extension 425, the call is connected and audio is available in both directions, then after 10 seconds audio is dropped in both directions.

Thanks,
Matt

[hubtech]

Internal calling is fine. When dialing am outside number there is no audio on either side. If I call the extension externally it's silent for a while then voicemail picks up. 

[Anonymous]

If you can get another debug with the instructions from my last post and upload it to the FTP server I would be happy to take a look.  I would need to see a debug for both of those problems separately (a separate debug file for each) along with a brief call flow description including the phone numbers and extensions used, which you can include in a separate text file in the .zip.

Thanks,

Matt

[Anonymous]

I went ahead and flagged this post as "Assumed Answered". If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons. This will make them visible and help other members of the community find solutions more easily. If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Thanks,

Matt