 2 edits | problems calling with voip and kx-tgp500 hello
this was initially under another topic, »voip is digital - why use an ata? but perhaps it should have it's own here.
i am having trouble setting this device up. actually, i think the configuration is ok at this point. i have both anveo and voip.ms going and they are registered just fine.
however, calling is another matter.
i am using a cable modem with a zywall usg20 firewall. my whole lan is plugged into p1 and the kx-tgp500 is plugged into p2. i changed the firewall rules to allow the non-standard 5010 port from anveo to pass thru. that seems to work. i'm getting both tx and rx packets on p2.
here is the problem tho: when i make calls (using either voipms or anveo) the other phone rings fine, but when they pick up, they get silence. however, *my* end keeps indicating the other end is still ringing.
calls in thru my voipms or anveo number don't ring here. the voipms doesn't even give me the ringing tone in the handset.
in both voipms and anveo, i can see cdr traffic both ways is being registered correctly in *their* servers. so i guess the problem is mine.
do i have a config problem with my firewall? -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 Reviews:
·Future Nine Corp..
| It sounds like it could be a NAT or network address translation problem. It looks like your ip phone supports a STUN server and if you don't already have it configured, I would configure a STUN server to help your IP Phone determine its external ip address and port numbers which it includes in the call sip signalling. You can use any STUN server such as stun.callwithus.com. |
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 | i have the tgp500 configured with stun.callwithus.com i checked my firewall right after this and noticed access denied. set up an access group for ports 5060 and 5070. the firewall then didn't block those ports but the incoming still doesn't ring.
i suspect the firewall, at this point, but it's not saying that it is blocking anything now. -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | voip.ms tech told me:
"If your router includes a SIP ALG and/or SPI Firewall setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers."
does anyone know how to disable the spi setting on the usg20? -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 PX EliezerPremium join:2008-08-09 Hutt River kudos:13 Reviews:
·callwithus
·voip.ms
·Optimum Voice
·Vitelity VOIP
·Gizmo5
| OMG a Zywall.
They have a reputation of talking a good game when it comes to VoIP, but being a PITA in real life.
Like Lindsay Lohan, always promises to behave but keeps getting into more trouble.
Here is the link to the manual, warning it is 50 MB! »ftp://ftp2.zyxel.com/ZyWALL_USG_20/use···_Ed1.pdf
BUT it won't open without a password!
This Zyzel company actually makes Cisco look good. |
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 1 edit | oooo*K*
so i just took the zywall out of the loop (73 minutes later) and my airport extreme is back to being my dhcp. the tgp500 is now just another thingy attached to the back of the airport.
the 500 registers just fine. the config looks good (as it was).
however, the call behavior is exactly as before. calling *in* to my voip.ms number, the handset is not ringing and the 500 doesn't ring. calling in to my anveo number gives me a ringing sound on the calling phone, but again the 500 doesn't ring.
the voip.ms tech said to ensure the SIP ALG is disabled. anyone know how to do this on an airport extreme? -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 Dan_voip join:2007-01-03 Saint-Hubert, QC kudos:3 | Compare your settings with the ones from this post. If doesn't work post some screen shoots. |
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 | there are minor differences in the sip settings. the voip settings are identical (except for the order of codecs) i have *nothing* set for the call control. and my japanese is kinda rusty .. but i took my firewall out of the loop anyways, so it's not a factor.
i'm concentrating on just voip.ms as of now since their tech support has been responsive. (anveo has maintained radio silence thru this process). i can make the minor changes to the sip settings. how do i configure the call control for voip.ms?
-- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 PX EliezerPremium join:2008-08-09 Hutt River kudos:13 Reviews:
·callwithus
·voip.ms
·Optimum Voice
·Vitelity VOIP
·Gizmo5
| On the Panasonic page entitled "Call Control" you really should not have to change anything.
If you are using Voip.MS then see: »wiki.voip.ms/article/Panasonic_KX-TGP_550
Note that you may need to download a special provisioning file:
The default registration expiry time on this device is 3600 seconds. You may have to reduce that if you lose registration. Unfortunately, that can not be set on the web interface. You need to load a provisioning configuration file from a web or FTP server....
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My Panasonic is easily set up for CallCentric, CWU, and Junction Networks' GetOnSip.
You want to use your Panasonic with Anveo and/or Voip.MS
I wish I could help more, but we are like ships passing in the night.  |
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 | reply to horacebork said by horacebork:there are minor differences in the sip settings. the voip settings are identical (except for the order of codecs) i have *nothing* set for the call control. and my japanese is kinda rusty .. but i took my firewall out of the loop anyways, so it's not a factor.
i'm concentrating on just voip.ms as of now since their tech support has been responsive. (anveo has maintained radio silence thru this process). i can make the minor changes to the sip settings. how do i configure the call control for voip.ms?
Following are my settings:
Phone Number: 100000 - Six-digit account number
Registrar Server Address: city.voip.ms - replace city with the city you are using Registrar Server Port: 5060 Proxy Server Address: city.voip.ms Proxy Server Port: 5060 Presence Server Address: city.voip.ms Presence Server Port: 5060
Service Domain: city.voip.ms
Source Port: 5090 (Line 4 in my case)
Authentication ID: 100000 - Six-digit account number Authentication Password: secret
Keep Alive Interval: 60
Hope this helps. -- DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 1.8.11.0 with Asterisk GUI on Virtual Server Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI |
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 3 edits |  kx tgp500 line 2 sip config (voip.ms) |  kx tgp500 line 1 sip config (anveo) |
grand total
i have almost all the same settings. i added the presence server and changed the keep alive to match yours. the service port, however, cannot be changed. i set it to 5090 and get 'failed' when clicking save. this is line 2. would 5070 suffice?
the rest seem ok. see screen snap. i'm lost
is SIP ALG disable something i need to attend to?
[**edit**] all seems to be working now. i have re-uploaded screen shots on this post to show both my current working voip.ms and anveo sip configs.
-- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | reply to horacebork well, i got anveo working inbound and outbound. seems i got the source port set to 5010 (overrode the 5060) for line 1. set it back to the correct 5060 and it works.
hip hip.
now for voip.ms (if i can do it). tomorrow. -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | reply to horacebork said by horacebork:...the service port, however, cannot be changed. i set it to 5090 and get 'failed' when clicking save. this is line 2. would 5070 suffice?
the rest seem ok. see screen snap. i'm lost
is SIP ALG disable something i need to attend to?
5070 for source port is fine. I'm surprised you can't change it.
For Enable DNS SRV lookup you *must* set 'No' for VoIP.MS.
If your router has it then you should disable SIP ALG. -- DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 1.8.11.0 with Asterisk GUI on Virtual Server Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI |
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 1 edit | grand total
the service ports on the tgp500 seem to be sequential. line 1 is 5060, 2 is 5070 etc. but apparently can't be duplicates. when i changed it to 5090 to set it to the same as yours, there already *was* a 5090. when i changed it to 5030 (as a test) the system allowed it. this is what got me suspicious in the first place.
for line 1, the default service port is 5060, which is the same as the default sip port 5060. when i went thru the anveo setup changing 5060s to 5010s, this one got erroneously thrown into the mix. so i changed it from 5010 back to 5060 and anveo started working (line 1).
then on line 2 (voip.ms) i had the register and proxy server set but not the presence. when i made my changes per your recommendation, actually it started working. however, i dialed the wrong number three times so it didn't ring my phone here and seemingly the symptoms were identical. i realized my mistake when i got a call from the person i called three times ... oops.
upshot? based on your information and how i applied it, both anveo and voip.ms are working correctly. i have updated an earlier post to show the latest config for both voip.ms and anveo working configs.
and yes: all the problems were pilot error. configs were wrong. i want to try putting the zywall back in to see if things will still work or not. now that i understand the port issues a bit better, maybe it'll be ok.
**thank you all** for the help in getting over the hurdle. -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | reply to horacebork next on the list is caller id. someone from anveo says they don't support outgoing caller id. really? so when i call, nobody gets a name? just a number?
not sure about voip.ms yet how they do (or don't) outgoing caller id. anyone with experience using voip.ms outgoing caller id? -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 PX EliezerPremium join:2008-08-09 Hutt River kudos:13 Reviews:
·callwithus
·voip.ms
·Optimum Voice
·Vitelity VOIP
·Gizmo5
| said by horacebork:next on the list is caller id. someone from anveo says they don't support outgoing caller id. really? so when i call, nobody gets a name? just a number? That [is] outbound Caller ID [CID].
You are seeking outbound Caller ID with Name [CNAM].
And with Anveo and others BTW that issue is a little different in Canada vs. the USA.
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Regarding outbound CNAM on Voip.MS, they may be able to provide that (which is actually LIDB populating) for a fee, you should ask them. That is, it is available on some lines but not others.... |
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 3 edits | px -- sorry for my bad terminology. i'm getting there.. right: caller id is the number cnam is the name.
anveo doesn't support cnam - i guess they won't update the cname database? they are based in canada, right? what does that have to do with it?
just heard back from voip.ms. they do permit cnam, for $10 (i *think* that's per DID) your statement about their cnam is almost exactly what he told me.
actually the tgp500 has a call control display name. but the only place i see that name is on the voip.ms cdr report. the anveo report doesn't show the name at all. in neither case does this name show up on the receiving phone. -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | reply to grand total grand total,
i just had an anveo tech (not voip.ms) tell me that for DNS SRV, i should set it to 'YES' and blank out the *port* for the registrar and proxy.
he said the port should be dynamically detected. sounded odd to me. however, the point was moot because the tgp500 does not allow ports to be set to /blank/. so.
for both services, i have the DNS SRV set to NO.
i still don't see any way to set SIP ALG on the airport extreme. -- ".. the sofa has just vanished." ".. well, that's one mystery less." |
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 | Anveo DNS SRV = Yes VoIP.MS DNS SRV = No
I was answering for VoIP.MS - definitely No for DNS SRV. If I had been answering about Anveo the answer would have been Yes. |
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 | gt:
then why does dns srv = NO work with anveo for my setup? |
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