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- About Dan Lukes
Dan Lukes
Member since
09-20-2011
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4256
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1142
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Created by
Dan Lukes
in Small Business Voice and Conferencing
in Small Business Voice and Conferencing
04-21-2019
03:20 AM
04-21-2019
03:20 AM
my phone connected to my modem via the SPA112. Whenever I call a place that has an automated message that says "press 1 for xxx, press 2 for yyy or press 3 for zzz" pushing the number does nothing it appears that the analogue phones signal does not get relayed correctly. Your phone is connected to your VoIP provider via SPA112. Your modem, internet connection type and even phone model is not important for the purpose of issue you described. The "signals" in question are known as DTMF. You are true, DTMF needs to be relayed correctly from your phone to your VoIP operator. There are multiple ways how to relay them. SPA112 needs to be configured to the method supported by your VoIP operator. It's nothing we can guess or analyze - just configure whatever method yours VoIP operator recommends to you. As a workaround/last resort solution, you ca use in-band type of relaying, but note the in-band is incompatible with compressed codecs, so use aLaw or uLaw (according your VoIP operator preference or according the native codec used in your country). This workaround may not work reliably, may not work to some destinations, or may not work at all. Correct relaying method based on VoIP operator advice is preferred solution.
... View more
Created by
Dan Lukes
in ATAs, Gateways and Accessories
in ATAs, Gateways and Accessories
04-20-2019
01:21 AM
04-20-2019
01:21 AM
QoS Policy: Always On On When Phone In Use Upstream Bandwidth: 10000 kbps I was just wondering if there was a way to turn it off, as the GUI doesn't offer that option. No, it's not possible. And I'm almost sure you wish not to turn it off. If you upstream line is not overloaded by data transfer during call, QoS have no effect, so it doesn't harm to have it on. If you upstream line IS overloaded, you wish the call audio still pass the line. It's the QoS is here for (assuming the other side of the line honor QoS as well) and you wish not to turn it off.
... View more
Created by
Dan Lukes
in ATAs, Gateways and Accessories
in ATAs, Gateways and Accessories
04-19-2019
12:57 PM
04-19-2019
12:57 PM
Is there a way to access your internal network using your private IP with a port? You didn't mention where you wish to access such target from. It is possible as long as source network knows how to route packets into destination. Its general rule - it has nothing to do with particular type/model of router you are using. Is there a way to turn off the QoS? What you mean "QoS" in this context ? May be you should describe the goal to avoid confusion.
... View more
Created by
Dan Lukes
in Voice Systems
in Voice Systems
04-19-2019
05:29 AM
04-19-2019
05:29 AM
Certificates are here to verify the phone is speaking to correct server (e.g. there's neither man-in-the-midle kind of attack or other unauthorized server trying to cheat phone with a rogue configuration). The list of trusted CA is critical part of it - if client (phone) trusts certificate regardless issuer, then it will accept unauthorized server (with certificate issued by any CA, even the rogue user own one). Phone shall reject a server with certificate issued by unknown CA. I didn't verified 7.6.x firmwares, but in 7.5.x it works as expected (with one exception - SPA5xx ignores certificate validity, e.g. "not before" and "not after" dates).
... View more
Created by
Dan Lukes
in Voice Systems
in Voice Systems
04-18-2019
07:14 AM
04-18-2019
07:14 AM
If such certificate has been accepted with no Custom CA defined, then you discovered severe security bug. in SR4. It seems it has been patched in SR5. It's unfortunate Cisco decided not to mention it in Release Notes. Cisco seems not to take security seriously ...
... View more
Created by
Dan Lukes
in Voice Systems
in Voice Systems
04-17-2019
02:25 PM
04-17-2019
02:25 PM
@Alex A. Welzl wrote: 7564 120.866512328 172.16.11.37 172.16.11.99 Syslog 63 LOCAL3.DEBUG: ssl cert err 19 err 19 is X509_V_ERR_SELF_SIGNED_CERT_IN_CHAIN self signed certificate not recognized as trusted depth=2 C = US, O = "Entrust, Inc.", OU = See www.entrust.net/legal-terms, OU = "(c) 2009 Entrust, Inc. - for authorized use only", CN = Entrust Root Certification Authority - G2 Why you think the Entrust Root Certification Authority - G2 should be accepted as trusted authority ? Documentation mentions no such CA on embedded list of trusted CA. OK. Did you configured such CA as Custom CA ? If yes, did you verified it's still configured even after firmware upgrade ? If no - it works as expected, the CA should not be trusted,
... View more
Created by
Dan Lukes
in Voice Systems
in Voice Systems
04-16-2019
11:43 PM
04-16-2019
11:43 PM
Extended feature of SPA50x configuration ? It can't be used to stream DTMF sequence to active call. Use standard numeric keypad to fire DTMF code.
... View more
Created by
Dan Lukes
in IP Telephony and Phones
in IP Telephony and Phones
04-16-2019
10:25 PM
04-16-2019
10:25 PM
Cisco 7841 phone Factory Reset using the Keypad You should consider to read Administrators guide of the device you are using (just advice) ...
... View more
Created by
Dan Lukes
in Voice Systems
in Voice Systems
04-16-2019
09:30 AM
04-16-2019
09:30 AM
It's not possible, as far as I know. SPA50x can trigger transfer of already connected call, but it will ask for target number. Transfer to voicemail initiated by a DTMF code may be supported by FreePBX itself. Consult FreePBX documentation.
... View more
Created by
Dan Lukes
in ATAs, Gateways and Accessories
in ATAs, Gateways and Accessories
04-14-2019
10:41 AM
04-14-2019
10:41 AM
you can use english or italian Not here. This is English community. I have a very easy configuration: adsl > router > SPA122 > phone. I don't claim it easy. Assuming the "ADSL" is "ADSL with router" (most common case) and assuming both routers are doing NAT you have two routers with NAT on the path (not counting the SPA122 is router as well). It's definitely not easy configuration. Both NATs and/or SPA122 needs to be configured properly. My problem is I don't receive call in but always I can phone out. OK, you need to understand how the SIP works for incoming call. SPA122 sends REGISTER message to the VoIP provider. Such REGISTER message inform VoIP provider where incoming calls shall be send to (IP and port). SPA122 knows nothing about two NATs on the path - it just claims own IP and a port as incomming call entry point. Unfortunately, this particular IP and port is unreachable from outer world - it's and private local address behind NAT. NAT engine running on router could catch REGISTER message and replace IP:port address inside by an outer (reachable) pair of IP:port maintaining mapping between such (outer) IP:port pair and original (SPA122 requested) IP:port pair. Contact your VoIP provider for support. Verify what IP:port address is registered on behalf of your phone. It must be outer IP address reachable for your VoIP provider. Also verify your firewall (either on ADSL router on an unspecified second router) doesn't block incoming calls. Note that working REGISTER is not required for outer calls.
... View more
04-10-2019
02:40 AM
It must be a misunderstanding. SPA8000 is 8-port analog terminal only. It have no other purpose but connecting up to 8 analog lines/phones. It doesn't support SIP digital phone of any kind. Either you have no SPA8000 (you may have SPA9000 instead) or your digital phones have nothing to do with it. Or the phones you are claiming to be digital are analog ...
... View more
04-10-2019
01:16 AM
We have a Cisco Linksys Sipura SPA 8000 voip port ip telefone gateway. The voip phone are working fine with it, but i have to connect some old analog phones to the systems. Sorry ? SPA8000 is 8-Port IP Telephony Gateway. It allows connections between eight analog telephones and an IP-based data network. So I don't understand yours "voip phone are working fine with it, but i have to connect some old analog phones". If you have SPA8000, you have solution - it's what the SPA8000 is designed for (but see notice bellow). Cisco SPA 112 adapter capable to convert the digital data to analog There's nothing like "standard analog phone". Analog phones you have assumes analog (subscriber) line of some specification. SPA112 creates subscriber (analog) line of some specification. It will work as long as both specifications are compatible. Compare yours phone requirements against SPA112 specification: Subscriber line interface circuit (SLIC) to answer your question by self. Or just try it ...
... View more
Created by
Dan Lukes
in ATAs, Gateways and Accessories
in ATAs, Gateways and Accessories
04-08-2019
06:14 PM
04-08-2019
06:14 PM
You need to have/provide syslog server. Value of Syslog server is the IP of such server. Read Debug and syslog Messages from SPA1x2 and SPA232D ATA
... View more
Created by
Dan Lukes
in Small Business Support Documents
in Small Business Support Documents
04-08-2019
06:08 PM
04-08-2019
06:08 PM
This document is attempt to recreate content of original document created by famous @Patrick Born. Cisco has considered to destroy such valuable document for an unknown reason. In order to capture debug and syslog messages from your SPA1x2 and SPA232D phones you need to enable debug and or syslog and either have a syslog server on your network or Wireshark running. [Wireshark will display debug/syslog messages with the "syslog" output filter]. Turn on debugging and or syslog messaging on your phone as follows [You don't need both debug and syslog, debug is sufficient if you are capturing data for support]: Determine the IP address of your syslog server (f or this example, assume a syslog server IP address of 192.168.0.200) Configure your phone by directing your browser to http://<IP_address_of_Phone>/ Example: http:// 192.168.0.100 / if the phone's IP address is 192.168.0.100 Log in as administrator Enable Debug Messages by editing the following fields: i. Phone's web-ui > Voice tab > System > Debug Server: <IP address of debug server> Example: 192.168.0.200 [if you are capturing with Wireshark, you do not need to use a valid syslog server's IP address, any "fake" address can be used, example 10.0.0.1] ii. Phone's web-ui > Voice tab > System > Debug Level: 3 iii. Phone's web-ui > Voice tab > Line N > SIP Debug Option: full Enable Syslog Messages by editing the following fields [this is not needed if you have enabled Debug]: Phone's web-ui > Voice tab > System > Syslog Server: <IP address of syslog server> Example: 192.168.0.200 [addresses of syslog and debug server can be same] The Phone will reboot and send all debug / syslog messages to the specified IP address of the syslog server. [Don't forget to make sure that your syslog server is running and that its IP address is what you configured on the phone.] Enable Syslog&Debug Messages Set SIP Debug Option For SPA3xx, SPA5xxG, SPA9xx, & WIP310 IP phones see Debug and syslog Messages from the SPA3xx, SPA5xxG, SPA9xx, & WIP310 IP Phones
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Created by
Dan Lukes
in ATAs, Gateways and Accessories
in ATAs, Gateways and Accessories
04-08-2019
04:39 PM
04-08-2019
04:39 PM
It mean the VoIP operator is sending CEST time, but claim it as GMT. In short - it sends wrong time. Bad news for you. PBX is considered critical (and trusted) component of VoIP. SPA112 will just honor the date/time provided by it. And PBX administrator have no reason to maintain incorrect time - so complain to it ...
... View more
6 hours ago
my phone connected to my modem via the SPA112. Whenever I call a place
that has an automated message...
yesterday
QoS Policy: Always On On When Phone In UseUpstream Bandwidth: 10000
kbpsI was just wondering if ther...
Friday
Is there a way to access your internal network using your private IP
with a port? You didn't mention...
Friday
Certificates are here to verify the phone is speaking to correct server
(e.g. there's neither man-in...
Created by
Dan Lukes
in Voice Systems
Thursday
Thursday
If such certificate has been accepted with no Custom CA defined, then
you discovered severe security...
Created by
Dan Lukes
in Voice Systems
Wednesday
Wednesday
@Alex A. Welzl wrote:7564 120.866512328 172.16.11.37 172.16.11.99 Syslog
63 LOCAL3.DEBUG: ssl cert e...
Tuesday
Extended feature of SPA50x configuration ? It can't be used to stream
DTMF sequence to active call. ...
Created by
Dan Lukes
in IP Telephony and Phones
Tuesday
Tuesday
Cisco 7841 phone Factory Reset using the KeypadYou should consider to
read Administrators guide of t...
Tuesday
It's not possible, as far as I know. SPA50x can trigger transfer of
already connected call, but it w...
Created by
Dan Lukes
in ATAs, Gateways and Accessories
Sunday
Sunday
you can use english or italianNot here. This is English community. I
have a very easy configuration:...
2 weeks ago
It must be a misunderstanding.SPA8000 is 8-port analog terminal only. It
have no other purpose but c...
2 weeks ago
We have a Cisco Linksys Sipura SPA 8000 voip port ip telefone gateway.
The voip phone are working fi...
Created by
Dan Lukes
in ATAs, Gateways and Accessories
2 weeks ago
2 weeks ago
You need to have/provide syslog server. Value of Syslog server is the IP
of such server.Read Debug a...
2 weeks ago
This document is attempt to recreate content of original document
created by famous @Patrick Born. C...
Created by
Dan Lukes
in ATAs, Gateways and Accessories
2 weeks ago
2 weeks ago
It mean the VoIP operator is sending CEST time, but claim it as GMT. In
short - it sends wrong time....
Public Statistics
| Date Registered | 09-20-2011 11:18 AM |
| Date Last Visited | |
| Total Messages Posted | 4,256 |
| Total Helpful Votes Received | 1142 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 10 | |
| 5 | |
| 5 | |
| 30 | |
| 5 |
.jpg "VIP Mentor"){kind=icon}
Helpful Votes From
Awards
Cisco Designated VIPs
2018 Small Business
Cisco Designated VIPs
2017 Small Business
Cisco Designated VIPs
2016 Small Business
Cisco Designated VIPs
2015 Small Business Voice ATA and SPA
Cisco Designated VIPs
2014 Small Business Voice - ATA and SPA
Community Spotlight Award
Small Business February 2013
Community Spotlight Award
Member's Choice, January 2019
