There seems to be a difference in the configuration or call handing with our polycom phones and adtran phones. Our adtran phones are able to call extensions on our other adtran system but our polycom phones on one side are not able to hear the audio when calling the other system. I've tested both types of phones on the same port and had no issues with the adtran phones.
Has anyone had this issue before or could someone recommend the settings/ftp files that need to be compared?
I've spent a lot of time on this issue and have compared polycom files with between our other 7100. I've matched them up and they seem to be good.
All polycom phones are IP332 and adtran are 712.
I found that enabling the polycom phones as simple remote phones fixed the one way audio to the other nv7100 on the LAN. The polycom phones on the other nv7100 are not enabled but still work.
I don't have a the firewall enabled on either NV7100 that are connected via LAN so I'm wondering why this fixed the audio problem. Anyone know what the remote phone setting changes in the header or why this works?
The adtran phones don't require remote-phone. Does anyone know why a polycom phone requires this setting but not an adtran phone?
You should not have to set the Polycom phones up as simple remote phones to make this work. My first suggestion is to check and make sure that media anchoring is off globally. It should be set to no ip rtp media-anchoring in the configuration. If you still have problems after that it would be helpful to know the topology of how these units are connected. It could be something related to a firewall or NAT device causing the problem, but I would expect the ADTRAN phones to operate the same way.
Can you recreate the issue and get a debug voice verbose, debug sip stack messages, and a debug isdn l2-formatted from the site with the problem? You can omit the isdn debug if you are not using a PRI. Once you have that debug please make a .zip file with it and the current configuration of both units and upload it to our FTP server with the instructions below:
Open Internet Explorer web browser
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
Reply to this post with the file name so we can retrieve it
Thanks a lot Matt.
The firewalls are turned off on the two 7100s as they are both behind a 908e on the same LAN. I ran the no ip rtp media-anchoring on the two 7100s. The RTP media-anchoring wasn't enabled prior either.
Thank you for the information. Can you supply another debug with debug voice verbose and debug sip stack messages both enabled at the same time for each unit? Ideally I would like to have a debug file from each unit for the same failed call containing the output from both of those debugs mixed together. Sorry, I did not clarify that earlier. Also, which user experiences the one way audio (6119 or 6400)?
Thanks for supplying the new debugs.
The first thing I would recommend is upgrading the Polycom 331 firmware. It is currently running 3.1.3 and should be on 3.2.7. The phone firmware requirements are outlined in the release notes for each version. You are running AOS R10.6, so this can be found on page 6 of the . The Polycom firmware can be downloaded from this page on our main website:
This may very well fix the issue, so please try this first. If you continue to have problems afterwards I would need the questions below answered:
You mentioned the problem was calling from a Polycom phone to another phone on the remote 7100, and that this works if the same call flow is recreated with an ADTRAN phone. The debug indicates extension 6119 is calling 6400, which is the voicemail extension number on the remote 7100 instead of another phone. Can you confirm if 6119 can call an extension that goes to another IP phone off the remote 7100?
Both units are on the same LAN but they have different default routes. One is to 192.168.0.23 and the other is to 192.168.0.15. I assume that the Polycom and ADTRAN phones off of each 7100 are using DHCP to obtain their default gateway so they should be the same (the voice vlan IP of each local 7100), but wanted to make sure these phones are not manually programmed. To confirm there is not a routing issue can you plug in a PC off of each 7100, set it to be in the voice vlan to mimic a phone, and then ping the a phone in the remote voice vlan as well as the 7100 sip trunk address?
For comparison purposes it would also help to see a debug of this working with the ADTRAN phones and if possible a packet capture when the issue is recreated again from the port the phone is connected to with matching debugs on each unit again.
It appears that the updated firmware fixed the one way audio issue that was happening with the polycom phones. I thought that the phones were running this version since I had put the files on the cflash but I didn't do this correctly.
I have noticed that there is an issue with system audio not playing or being heard when calling from a polycom to the other 7100. By system audio I mean that there is no audio with the following, ring tone, voicemail greetings, auto attendant, call queue, and hold music. I removed the simple remote phone option. I was using the voicemail extension as a test and this caused confusion since the other audio appears to be fixed by the phone firmware update. If I set the simple remote phone option it will allow the polycom phone to hear the system audio.
I did a quick comparison between a new adtran phone call debug and a polycom phone on the far end 7100. I noticed that a=silenceSupp:off - - - - is present along with a=fmtp:18 annexb=no for the adtran phone. I don't know if this was a configurable option somewhere or if the polycom phone doesn't support this option. A guess is that RTP traffic isn't being sent. Is VAD configurable? The comparision is shown below:
far end debug with adtran phone:
19:31:52.907 SIP.STACK MSG v=0
19:31:52.908 SIP.STACK MSG o=MxSIP 0 1039271676 IN IP4 192.168.5.30
19:31:52.908 SIP.STACK MSG s=SIP Call
19:31:52.908 SIP.STACK MSG c=IN IP4 192.168.5.30
19:31:52.908 SIP.STACK MSG t=0 0
19:31:52.908 SIP.STACK MSG m=audio 3000 RTP/AVP 0 18 101
19:31:52.908 SIP.STACK MSG a=ptime:20
19:31:52.909 SIP.STACK MSG a=sendrecv
19:31:52.909 SIP.STACK MSG a=silenceSupp:off - - - -
19:31:52.909 SIP.STACK MSG a=rtpmap:0 PCMU/8000
19:31:52.909 SIP.STACK MSG a=rtpmap:18 G729/8000
19:31:52.909 SIP.STACK MSG a=fmtp:18 annexb=no
19:31:52.909 SIP.STACK MSG a=rtpmap:101 telephone-event/8000
19:31:52.910 SIP.STACK MSG a=fmtp:101 0-15
19:31:52.910 SIP.STACK MSG
19:31:52.912 TM.T04 01 SipTM_Idle rcvd SIP call-leg request: INVITE
19:31:52.913 TM.T04 01 SipTM_Idle call-leg -> Offering
19:31:52.913 TM.T04 01 SipTM_Idle State change >> SipTM_Idle->SipTM_Trying
with polycom phone:
00:50:17.730 SIP.STACK MSG v=0
00:50:17.730 SIP.STACK MSG o=- 1359611419 1359611419 IN IP4 192.168.5.14
00:50:17.730 SIP.STACK MSG s=Polycom IP Phone
00:50:17.730 SIP.STACK MSG c=IN IP4 192.168.5.14
00:50:17.730 SIP.STACK MSG t=0 0
00:50:17.731 SIP.STACK MSG a=sendrecv
00:50:17.731 SIP.STACK MSG m=audio 2236 RTP/AVP 0 18 127
00:50:17.731 SIP.STACK MSG a=rtpmap:0 PCMU/8000
00:50:17.731 SIP.STACK MSG a=rtpmap:18 G729/8000
00:50:17.731 SIP.STACK MSG a=rtpmap:127 telephone-event/8000
00:50:17.732 SIP.STACK MSG a=fmtp:127 0-15
Thank you for looking into the previous debugs and for your help. If I have to enable simple remote phones I think that this is an acceptable solution but was worried there was a bigger problem.
I'm glad to hear the new firmware fixed the issue. For the system audio problem I would need to see the same debugs. It would help to see them not working with a Polycom phone with an accompanying packet capture if possible, and then working with an ADTRAN phone for comparison purposes (no packet capture necessary for the working case).
I've uploaded GPtoVoicemailDebug-PacketCapture.zip.
I noticed a change in the symptoms after I restarted both 7100s. The adtran phones are now behaving the same as the polycom phones. When calling from an adtran or polycom I don't hear the voicemail greeting, auto attendant, or call queue music. I upgraded the firmware on the adtran phones to match the supported firmware in the documentation. The upload contains the debugs from both units and a packet capture from the far end unit. I let the call in the uploaded capture go to voicemail. I don't know if the correct debug is being used to troubleshoot this issue. If the phone is answered audio works fine so I don't understand why other audio that is played from cflash doesnt come through properly. Frustrating problem.
Thank you for your work thus far Matt.