I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XXXXXX secret=XXXXX host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt
if your connection is also used for web, email, and the worst, p2p, you better to have qos on your router. just be aware that g711 will use 80Kb up and down... gsm and g729 wil use 30/40Kb then : disallow all allow = gsm allow = g729 Olivier kurt x a ?crit :> I have an Asterisk server that I use at work. I have a phone that is > at home that logs into > the Asterisk server at work. My home phone is hooked up via DSL > through a Linksys router. You can see the my sip.conf for the phone > blow. > > The problem is each time the phone rings I can hear/be heard 50% of > the time. > > Any suggestion on what to look for. > > I do have my reg time set for 180 seconds on the cisco ATA186. > > [72459] > type=friend > username=XXXXXX > secret=XXXXX > host=dynamic > context=voice-mail > dtmfmode=rfc2833 > ;canreivet=yes > nat=yes > qualify=yes > > Kurt > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
This is most likely your upload speed. I have Comcast supposedly with 384KB upload, but I have a hard time using VoIP unless I use a low-bandwidth codec like GSM. For g711, it's a crap shoot as to whether it works or not. I can always hear the other person clearly since I have a ton of download bandwidth available, but they have a hard time hearing me and I tend to break up a lot. Derek On 5/17/06, kurt x <kurtwp@gmail.com> wrote:> I have an Asterisk server that I use at work. I have a phone that is > at home that logs into > the Asterisk server at work. My home phone is hooked up via DSL > through a Linksys router. You can see the my sip.conf for the phone > blow. > > The problem is each time the phone rings I can hear/be heard 50% of the time. > > Any suggestion on what to look for. > > I do have my reg time set for 180 seconds on the cisco ATA186. > > [72459] > type=friend > username=XXXXXX > secret=XXXXX > host=dynamic > context=voice-mail > dtmfmode=rfc2833 > ;canreivet=yes > nat=yes > qualify=yes > > Kurt > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I would agree, if I also experience choppy voice. Over the last month I had one spike of 893k over my T1. My average usually is 223k. I carved out 640k for voice QOS on the WAN router. At most I would have 4 calls up at once. The call comes in, the phone rings, 50% of the time I can have a conversations. 50% of time I can not. Maybe I should complain to my SIP service provider. Kurt ----------------------------------------------------------------------------------------------- if your connection is also used for web, email, and the worst, p2p, you better to have qos on your router. just be aware that g711 will use 80Kb up and down... gsm and g729 wil use 30/40Kb then : disallow all allow = gsm allow = g729 Olivier kurt x a ?crit :> I have an Asterisk server that I use at work. I have a phone that is > at home that logs into > the Asterisk server at work. My home phone is hooked up via DSL > through a Linksys router. You can see the my sip.conf for the phone > blow. > > The problem is each time the phone rings I can hear/be heard 50% of > the time. > > Any suggestion on what to look for. > > I do have my reg time set for 180 seconds on the cisco ATA186. > > [72459] > type=friend > username=XXXXXX > secret=XXXXX > host=dynamic > context=voice-mail > dtmfmode=rfc2833 > ;canreivet=yes > nat=yes > qualify=yes > > Kurt
A few things; You have nat and qualify = yes, those settings are correct. On your DSL, is there a public IP address on the internet side of the Linksys? (not in the 10.x.x.x, 192.168.x.x, or 172.16.x.x subnets). If not, you have another NAT router in the middle (your DSL modem) and you will not have good luck. The ATA186 is an antique in the voip world, it is no longer supported and has poor and primitive SIP firmware. Spend $80-90 and get a Sipura SPA2100. It is a combo router/ATA that works very well and has very refined SIP images. Use 3.2.5(d) with asterisk 1.2 for best results. Make sure you set the RTP packet size to .020 if you do use a SPA2100, the default is .030 (30ms). Make sure your Linksys router has current firmware. The newer (hardware version 4) 4 and 8 port wired routers support QoS, as well as the WRT54G. if you have one of those you should turn on the QoS, my guess is you do not or the issue you report would likely not exist. Try putting the ATA in the "DMZ" on the Linksys by setting the DMZ host, but make sure your ata is secured with strong passwords since it will become accessible from the internet. The issue you are having is most likely NAT/Firewall related. When you embrace newer technologies like VoIP you must also be willing to embrace the cost of modern hardware, and the ATA186 does not fall into that category... it was the device the Cisco/Linksys/Sipura folks cut thier teeth on, and like many other 1st generation products, it sucks. The same engineers that designed that device are still designing Cisco/Sipura/Linksys devices; they just have a lot more experience now. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kurt x Sent: Wednesday, May 17, 2006 12:37 PM To: Asterisk Subject: [Asterisk-Users] Audio problems 50% of the time. I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XXXXXX secret=XXXXX host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I get the impression the complaint is NO audio, not poor audio. This points more to NAT than QoS. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Lee-Wo Sent: Wednesday, May 17, 2006 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Audio problems 50% of the time. This is most likely your upload speed. I have Comcast supposedly with 384KB upload, but I have a hard time using VoIP unless I use a low-bandwidth codec like GSM. For g711, it's a crap shoot as to whether it works or not. I can always hear the other person clearly since I have a ton of download bandwidth available, but they have a hard time hearing me and I tend to break up a lot. Derek On 5/17/06, kurt x <kurtwp@gmail.com> wrote:> I have an Asterisk server that I use at work. I have a phone that is > at home that logs into > the Asterisk server at work. My home phone is hooked up via DSL > through a Linksys router. You can see the my sip.conf for the phone > blow. > > The problem is each time the phone rings I can hear/be heard 50% ofthe time.> > Any suggestion on what to look for. > > I do have my reg time set for 180 seconds on the cisco ATA186. > > [72459] > type=friend > username=XXXXXX > secret=XXXXX > host=dynamic > context=voice-mail > dtmfmode=rfc2833 > ;canreivet=yes > nat=yes > qualify=yes > > Kurt > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users