07-08-2009 07:16 AM - edited 03-21-2019 09:17 AM
Hi,
We have following situation. Asterisk as PBX and 2 SPA942 and 1 PAP2 with 2 analog phones connected. All is working prefectly until we added WRP400 to replace PAP2. Analog phones connected to WRP400 are registered on Asterisk. They are ringing on call but no audio on both sides. SPA 942 connected to WRP400 haven't any problem and PAP2 connected to the WRP400 with both analog phones is working perfectly but only voice piece of WRP400 isn't working.
We have checked configuration of voice piece of WRP400 many time and it seems all properly configured. An port forwarding or firewall issue couldn't be because SPA942 and PAP2 connected to the WRP400 are working fine. Firmware upgrade doesn't solved our problem.
Have you any idea what could be the problem? Any help would be appreciated.
P.S. Sorry for my bad english because isn't my mother language.
Best regards,
Pera
09-30-2009 04:07 AM
Hi Pera,
I need:
1. The voice configuration of the WRP400 please. Use these instructions: https://supportforums.cisco.com/docs/DOC-9906
2. To know, does your service provider use a session border controller (SBC)?
Thanks,
Patrick
----------
09-30-2009 06:21 AM
Hi Patrick,
1. Do you need WRP400 voice configuration with connected phonet to the WRP400 or just so as it because analog phones not use WRP400 but PAP-2.
2. Not a clue. We have 2 providers:
www.sipcall.ch
Betamax with many clones such Voipbuster, Poivy etc.
But the problem is between local phones too. Do you thins is voice part of WRP400 probably fault?
Regards,
Pera
09-30-2009 08:22 AM
Hi Pera,
I am under the impression that you have 2 analog phones connected to the WRP400 that ring, but when answered, do not produce audio. From the https://supportforums.cisco.com/message/3094931#3094931 description, the same analog phones work without problem in the same environment when connected to your PAP-2 instead of the WRP400.
Can you please restate the problem that we're trying to resolve? :-)
Regards,
Patrick
----------
09-30-2009 09:30 AM
> I am under the impression that you have 2 analog phones connected to the WRP400 that ring, but when answered, do not produce audio. From the > https://supportforums.cisco.com/message/3094931#3094931 description, the same analog phones work without problem in the same environment when > connected to your PAP-2 instead of the WRP400.
Exactly. And our PAP-2 is connected to the WRP400. And no audio is fi I call or be called from or analog phones.
Regards,
Pera
09-30-2009 02:55 PM
Pera,
Is the analog phone connected to PHONE 1 or PHONE 2 of the SPA400?
Regards,
Patrick
----------
10-01-2009 12:25 AM
Hi,
I haven't had a look on your configs yet, but maybe I'm hitting a point:
Just to be sure that the audio part of the WRP400 is not broken anyhow, is it possible for you to access the IVR by entering ' * * * * ' on any phoneport? The device will confirm every key in the IVR by repeating it phonetically. If that is working, you can assume your device being not broken (never say never :-)
BTW, I suggest adding an echo extension to your asterisk dial plan in extensions.conf, e.g.
exten => _01189876!,1,Echo() ; use _0089876! in Europe
Try to dial that number and you should hear yourself. Do you?
Another issue: is your Asterisk connected to one of the LAN ports of the WRP400 and do you have STUN and NAT enabled on the phone lines? That will probably not work! I just roughly inspected the configs. STUN does not seem to be enabled. You are using FreePBX. Changing the configuration files manually there is hard work if you're not in it, well I'm not :-)
Best regards,
Eik Rentzow
10-01-2009 01:17 AM
Echo test allready exsist on the Asterisk. Good idea. I'l try.
Asterisk isn't connected to the WRP400 but to the another switch in LAN and STUN isn't used anyway.
10-01-2009 12:32 AM
I have 2 analog phones. One analog phone connected to PHONE 1 and other analog phone connected to PHONE 2 of the WRP400 (not SPA400).
Last try was with only one analog phone connected to PHONE 1 as I made phone configuration picture.
10-01-2009 04:33 PM
Hi Pera,
Sorry about the typing error, I really did mean WRP400... too many products with similar names.
Thanks for confirming that the analog phones are connected to the WRP400.
Please supply the WRP400 config for analysis: https://supportforums.cisco.com/message/3094946#3094946
Regards,
Patrick
----------
10-02-2009 01:51 AM
10-22-2009 02:37 PM
Hi Pera,
Apologies for the delayed response, I've been on business travel.
One thing to keep in mind is that a PAP2 is not the same device as a WRP400. The WRP400 is much more complex....
Looking at your config, I see that you have changed the SIP and RTP ports. A typical problem when RTP is restricted, in any way, is one-way audio.
Please consider factory-defaulting the WRP400 and configuring the bare minimum to get it working, as in SIP proxy and user credentials only.
Once you have verified that 2-way audio works, change the settings on the WRP400 in small increments, making sure that 2-way audio continues to function.
Consider using a combination of WRP400 syslog, Asterisk console messages, and Wireshark traces so you can immediately determine where the RTP stream is going wrong.
Best of luck,
Patrick
-----------
10-23-2009 01:08 AM
> Looking at your config, I see that you have changed the SIP and RTP ports. A typical problem when RTP is restricted, in any way, is one- > way audio.
I have NO AUDIO and not one-way audio. The ports are the same such in PAP2 and RTP ports from Asterisk config. I can this problem with one-way audio and RTP ports exactly and I tried allready without success.
> Please consider factory-defaulting the WRP400 and configuring the bare minimum to get it working, as in SIP proxy and user credentials > only.
>
> Once you have verified that 2-way audio works, change the settings on the WRP400 in small increments, making sure that 2-way audio > continues to function.
>
> Consider using a combination of WRP400 syslog, Asterisk console messages, and Wireshark traces so you can immediately determine where the > RTP stream is going wrong.
I see. You have not a clue. I configured many VOIP devices and have had no one problem only with this WRP400. And if I have so much knowledge to work with WRP400 syslog, Asterisk console messages, and Wireshark traces as you say then my question here was unnecessary because you mean I can solve it himself. For what I sent so much informations? If you see anything there to help me then tell me please.
I can give a try with factory-defaulting again but what if I have no audio furthermore? WRP400 in trash can? Probably is the best solution instead to waste time and no more to buy Linksys/Cisco products.
Pera
10-27-2009 07:48 AM
1. factory reset of WRP400
2. Line 1 only user credentials and proxy.
3. Attached analog phone on line 1 of WRP400
Trying to call another extensions form phone on line 1. I hear after dialing only busy tone. No activity in Asterisk log.
Trying from another extentions to call phone on line 1. The phone is ringing but after answering no audio (on both sides).
So your solution is nothing worth. And if WRP400 isn't defect it's very bad product.
10-28-2009 08:24 AM
Hi Pera,
I registered a factory defaulted WRP400 [running 1.01.00] Line 1 to my Asterisk server and can make calls with audio between an IP phone on the LAN and an analog phone attached to PHONE 1 of the WRP400.
I completed this task in about 5 minutes, without any trouble.
I did the following:
1. Edited Asterisk server:
a. sip.conf
b. extensions.conf
c. *CLI> sip reload
d. *CLI> dialplan reload
2. Configured WAN interface of WRP400
[it's a router, if this interface is not properly configured, the voice lines will not properly register with the Asterisk server]
a. Configured LAN inteface to static
b. WRP400 Line 1>
i. Proxy & Registration > Proxy > 192.168.2.245:5060 [IP address of asterisk server]
ii. Display Name> wrp400line1
iii. user ID: 401
iv. Password: 401secret
3. Submit all changes
4. I can now make calls with 2-way audio.
With VoIP, two protocols are used, SIP and RTP. SIP controls the session, but RTP carries the audio. The called phone ringing means that SIP is getting through. The lack of audio means that RTP is not getting through. This is why I encourage you to not change any SIP or RTP ports on the WRP400 until you have a basic configuration working.
As I mentioned in an earlier post, the WRP400 is not the same as a PAP2 device. It includes router functionality which increases the complexity of the solution as compared to using a PAP2.
Regards,
Patrick
-----------
11-12-2009 07:41 AM
> 2. Configured WAN interface of WRP400
> [it's a router, if this interface is not properly configured, the voice lines will not properly register with the Asterisk > server]
What you mean exactly with "[it's a router, if this interface is not properly configured, the voice lines will not properly register with the Asterisk server] "?
WRP400 isn't directly attached to the cable modem. It's Linksys BEFSX41 connected to the cable modem and act as router and WRP400 is connected via LAN to the Linksys BEFSX41. As I understand Admin Manual from here: http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/wrp400/administration/guide/WRP400_AG_OL-19688.pdf it schouldn't be any problem in this configuration or am I wrong?
I attached my basic setup of WRP400.
Regards,
Pera
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: