DWAyotte Posted October 16, 2009 Report Posted October 16, 2009 I have frequent complaints about calls that only 1 party can hear the other. What is the best way to troubleshoot these sorts of issues? I have looked through a packet sniff, but am not seeing anything that looks to be an issue. Perhaps my eyes are not trained to see the proper errors. I am wanting to get some pointers from you guys on how to best troubleshoot these sorts of issues. Thanks a ton. Quote
pbxuser911 Posted October 18, 2009 Report Posted October 18, 2009 is it intermitting? try on the admin/setting/port page set Lock codec during conversation: to yes do the same for /admin/domain/selectyourdomain/trunks/ Lock codec during conversation: set it to yes Quote
DWAyotte Posted October 21, 2009 Author Report Posted October 21, 2009 is it intermitting? try on the admin/setting/port page set Lock codec during conversation: to yes do the same for /admin/domain/selectyourdomain/trunks/ Lock codec during conversation: set it to yes This didn't seem to do the trick. I am still getting a handful of "I can't hear the caller" complaints each day. Any other ideas? I would love to be able to nail this down. Thanks, Quote
pbxuser911 Posted October 21, 2009 Report Posted October 21, 2009 what type of routers are they using? seems as there is issues on the users network, have you checked the network to assure there is no issues? Quote
andrewgroup Posted October 27, 2009 Report Posted October 27, 2009 could you provide details on the configuration of the environment? audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls. On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected. If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers. Quote
DWAyotte Posted October 27, 2009 Author Report Posted October 27, 2009 could you provide details on the configuration of the environment?audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls. On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected. If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers. my pbx sits on my LAN and on the internet, no NAT (2 NICs). I have all the necessary internet facing ports open and the rest closed (a total of 100 RTP ports). I have had 1 way on LAN more times then I feel comfortable having. So if it is my switches, what can I do? What 802.1x features should I be verifying? Quote
shopcomputer Posted October 27, 2009 Report Posted October 27, 2009 my pbx sits on my LAN and on the internet, no NAT (2 NICs). I have all the necessary internet facing ports open and the rest closed (a total of 100 RTP ports).I have had 1 way on LAN more times then I feel comfortable having. So if it is my switches, what can I do? What 802.1x features should I be verifying? How many calls, are you sure 100 rtp ports are enough for you? Quote
DWAyotte Posted October 27, 2009 Author Report Posted October 27, 2009 How many calls, are you sure 100 rtp ports are enough for you? not really sure no. what is the rule of thumb on rtp ports? The most concurrent calls I have had is 16. Quote
shopcomputer Posted October 27, 2009 Report Posted October 27, 2009 not really sure no. what is the rule of thumb on rtp ports?The most concurrent calls I have had is 16. How many phone do you have? Quote
DWAyotte Posted November 10, 2009 Author Report Posted November 10, 2009 How many phone do you have? 115 Quote
shopcomputer Posted November 10, 2009 Report Posted November 10, 2009 115 you are very low on the amount of rtp ports, at a minimum you need 2 per call and 1 per phone, increase your rtp ports to at least 1000. there is no benefit of limiting it, why not leave the default range. Quote
Vodia PBX Posted November 23, 2009 Report Posted November 23, 2009 When it comes to one-way audio check out this one: https://www.pbxnsipsupport.com/index.php?_m...ratingconfirm=1 It contains a small list of things that you can verify quickly. Quote
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