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This is a discussion on Sipgate and ATA186 within the uk.telecom.voip forums, part of the Newsgroup Forums category; PhilT wrote:[color=blue] > Malcolm Loades wrote: >[color=green] >> Unfortunately this guide is for MGCP not ...
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PhilT wrote:[color=blue]
> Malcolm Loades wrote: >[color=green] >> Unfortunately this guide is for MGCP not SIP.[/color] > > are you saying the audio bitmap does not exist on the SIP config ? > > Having run both MGCP and SIP there is a lot of common ground on the > settings, as is clear from the Cisco documentation on the web. > > The Sipgate page shows "0x00150015" as the "AudioMode:" setting. > > The default in the manual is 0x00350035 > > With my limited bitmap skills it appears that the Sipgate setting is > for "by negotiation" so I suggest you try 0x00050005 which is "DTMF > always in band". >[/color] Of course you will want this to be always outband if you use G.729. so presumably 0x00250025. |
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Malcolm Loades wrote:[color=blue]
> This works perfectly with the exception of not being able to send DTMF > tones over the line - dialling is no problem. Therefore I'm unable to > call most business lines which answer with "For xxxx please press y". > This also rules out voicemail. > > I've changed the analogue handset to an alternative which makes no > difference. The ATA186 is running Version: v3.1.0 atasip (Build > 040211A). > > Any ideas, please?[/color] This sounds exactly the same as the problem I have using the SMC7908VoWBRA router. DTMF tones are not recognized so I cannot use the Sipgate voicemail by dialling 50000. A few months ago I spent quite a lot of time emailing Sipgate tech support without resolving it. I seem to recall discovering that there are two varieties of out of band DTMF - perhaps my SMC and your device only implements one of them. I tried again today as I found out that there is a firmware upgrade for my device, but still no luck unfortunately. It's annoying, as it works great otherwise. |
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PhilT wrote:[color=blue]
> Thomas Kenyon wrote: >[color=green] >> Of course you will want this to be always outband if you use G.729. >> so presumably 0x00250025.[/color] > > depends whether outband works or not I suppose :-) > > Phil >[/color] True, but inband isn't supposed to work with that codec. (well, support is non-standard). |
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In message <1157491110.627420.121870@p79g2000cwp.googlegroups.com>,
PhilT <newsnet@gmail.com> writes[color=blue] > >Malcolm Loades wrote: >[color=green] >> Unfortunately this guide is for MGCP not SIP.[/color] > >are you saying the audio bitmap does not exist on the SIP config ? > >Having run both MGCP and SIP there is a lot of common ground on the >settings, as is clear from the Cisco documentation on the web. > >The Sipgate page shows "0x00150015" as the "AudioMode:" setting. > >The default in the manual is 0x00350035 > >With my limited bitmap skills it appears that the Sipgate setting is >for "by negotiation" so I suggest you try 0x00050005 which is "DTMF >always in band". >[/color] Thanks for the suggestions, and also everyone else who responded. My setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2 ...... still no DTMF detected by the other end. More Googling brings up lots of reports similar to mine but no solution :-( Unless someone knows better! Malcolm |
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Malcolm Loades wrote:[color=blue] > In message <1157491110.627420.121870@p79g2000cwp.googlegroups.com>, > PhilT <newsnet@gmail.com> writes[color=green] > >The Sipgate page shows "0x00150015" as the "AudioMode:" setting. > > > >The default in the manual is 0x00350035 > > > >With my limited bitmap skills it appears that the Sipgate setting is > >for "by negotiation" so I suggest you try 0x00050005 which is "DTMF > >always in band". > >[/color] > Thanks for the suggestions, and also everyone else who responded. My > setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2 > ..... still no DTMF detected by the other end. >[/color] 0x00050005 works fine with sipgate on an ATA186 calling out to a BT line, I can hear the tones and they work with Call Minder's IVR menu including accepting a PIN and navigating the menu. They don't seem to work with Sipgate's voicemail. Version: v3.1.0 atasip (Build 040211A) Phil |
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In message <1157895891.437427.317100@d34g2000cwd.googlegroups.com>,
PhilT <newsnet@gmail.com> writes[color=blue] > >Malcolm Loades wrote:[color=green] >> In message <1157491110.627420.121870@p79g2000cwp.googlegroups.com>, >> PhilT <newsnet@gmail.com> writes[color=darkred] >> >The Sipgate page shows "0x00150015" as the "AudioMode:" setting. >> > >> >The default in the manual is 0x00350035 >> > >> >With my limited bitmap skills it appears that the Sipgate setting is >> >for "by negotiation" so I suggest you try 0x00050005 which is "DTMF >> >always in band". >> >[/color] >> Thanks for the suggestions, and also everyone else who responded. My >> setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2 >> ..... still no DTMF detected by the other end. >>[/color] > >0x00050005 works fine with sipgate on an ATA186 calling out to a BT >line, I can hear the tones and they work with Call Minder's IVR menu >including accepting a PIN and navigating the menu. They don't seem to >work with Sipgate's voicemail. > >Version: v3.1.0 atasip (Build 040211A) >[/color] Superb! Thanks! I'd been doing the testing of each setting by calling Sipgate's voicemail - no wonder it failed. The setting you suggest does indeed work fine otherwise - I can live with that. Malcolm |
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Malcolm Loades wrote: [color=blue] > I'd been doing the testing of each setting by calling Sipgate's > voicemail - no wonder it failed.[/color] I'm afraid I have learned that... testing + sipgate = insanity + invalid conclusions the variability of the service is too great for robust testing. Phil |
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"PhilT" <newsnet@gmail.com> wrote in message
news:1157994927.745878.126390@h48g2000cwc.googlegroups.com[color=blue] > Malcolm Loades wrote: >[color=green] > > I'd been doing the testing of each setting by calling > > Sipgate's voicemail - no wonder it failed.[/color] > > I'm afraid I have learned that... > > testing + sipgate = insanity + invalid conclusions > > the variability of the service is too great for robust > testing.[/color] How so..? It works fine here and has done almost 100% since the server upgrades in April. Ivor |
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