 Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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I have changed the SIP port. Would know how it impact over next week usage. of my phone.
I was using two PlugLink 9650 ethernet adapters (they plug to power outlets and have ethernet port) to provide internet connection to my Cisco SPA 3102. My residential gateway was in one room and ATAs in different room. AT&T technician pointed out that I would be getting only 1.5Mbps speed from these adapters while my residential gateway is giving 12Mpbs. I was able to install a repeater bridge (flashing dd-wrt to wrt54g) to provide close to 8Mpbs to the ATAs. I am not seeing the call drop after this.
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| Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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Looking at my log:
Feb 14 00:06:26 VERBOSE[87791] logger.c: -- Registered SIP 'tjsingh-201' expires 300
Feb 15 00:18:20 VERBOSE[87791] logger.c: -- Registered SIP 'tjsingh-201' expires 300
It does seem that it is registering one per day. I have now set the registration interval to 60 sec.
I have a parallel discussion at ATT Uverse Forum:
http://forums.att.com/t5/Features-and-Ho...all/m-p/3426287
There was interesting following comment. It is to be seen if that makes a difference. I will keep you posted. Thanks for your continued interest.
I located this about port 5060:
"The U-Verse service requires the customer to use a router that was designed specifically for U-Verse Internet, TV, and phone service. This router has a default setting called SIP-ALG that cannot be disabled and the router cannot be replaced while using U-Verse internet. SIP-ALG is a router function that will cause VoIP traffic to be rewritten, which can cause problems such as one way audio, dropped calls, and inbound and outbound call failure. If you are familiar with making changes to your voice over IP phone/device, you will want to modify the source port of the phone to a port other than 5060 and reboot the U-Verse router."
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| Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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I made two outgoing US calls that dropped after 15 minutes and 29 minutes respectively. The battle is far from over. Need to find a way to debug this. Will contact future9 to see if they can help. PBX status log shows:
NOTICE[87791] chan_sip.c: Disconnecting call 'SIP/tjsingh-201-d87b' for lack of RTP activity in 61 seconds
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| Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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Hi Tel,
After resetting the SPA 2102 and PAP2T, problem was still there. I stumbled upon this post:
http://www.toao.net/25-linksys-ata-configuration
and the only change I made after the factory reset was
>>As we planned to place our device behind a router, we turned on NAT Mapping and NAT Keep Alive.
(I also changed SIP T1 to 1 and Reg Retry Long Intvl to 120. but I don't think it is related to dropped calls.)
I didn't mess with other NAT support parameters yet, which are No by default.. I have come across posts which talk about setting some of these parameters to Yes when NAT Mapping is enabled.
NAT Support Parameters
Handle VIA received: Handle VIA rport:
Insert VIA received: Insert VIA rport:
Substitute VIA Addr: Send Resp To Src Port:
STUN Enable: STUN Test Enable:
I was able to make outgoing local call (which uses future9 voip) that did not drop even after 15 minutes. Earlier it used to drop after 9-10 minutes. I will keep my finger crossed and see if it holds for incoming (future9 voip) and international calls (voipdiscount voip)
I also found that Future 9 also recommended NAT Mapping Enable:
http://www.future-nine.com/faq/content/1...spa-device.html
Will keep you posted.
Thanks for your support.
Regards.
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| Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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Thanks Tel!! I would give it a shot. I am also using Cisco PAP2T on another line and have similar issue. Should I try setting it to factory default also or does it need special network configuration? It is bit surprising that the I need to change the settings after switching the internet provider. It would nice to be able to debug just in case the problem persists.
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| Thread: RE: Need Sipura SPA 2102 Setting with ATT Uverse router |
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I swtiched from comcast to ATT Uverse and now the call drop after 5-6 minutes. I am using SIPURA SPA 2102 and my voip provider is future9.com for local calls and voipdiscount for international calls.
I am forwarding the port 5060 to my device IP address 192.168.1.101. The device is set to DHCP but I am using the router to reserve this address. I looked for configuration guide for SPA 2102 in the forum, but failed to find it.
Can you help?
I saw in system log:
NOTICE[8998] chan_sip.c: Disconnecting call 'SIP/tjsingh-201-cded' for lack of RTP activity in 61 seconds
Not sure if this message is useful.
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| Thread: RE: PAP2T/SPA2102: Can't call internal extension and incoming call fails |
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I flashed the DIR-601 router with DD-WRT firmware and I am in business.
http://www.dd-wrt.com/wiki/index.php/DIR-601
I can make inbound calls and also I am able to call the extensions internally. All issues resolved.
Thank a lot for your help for pointing out that this could be a router issue. I am now using the RESET config on both PAP2T and SPA2102.
I would setting a remote extension outside my LAN and test to see I can seemlessly call all extensions. Even if the extension is in India, it should work. right?
Best Regards.
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| Thread: RE: PAP2T/SPA2102: Can't call internal extension and incoming call fails |
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Thanks telagente00 for a quick reply.
I followed your instructions:
1. Audio Bypass is already "no" by default for me in PBXes extensions.
2. I did RESET of the PAP2T (73738# on IVR menu). So I have now factory default settings.
3. However when I turn-OFF SIP ALG (uncheck the box in Advanced->Firewall Settings), I can't make even the outbound calls.
This posts also suggests that SIP ALG should be turned ON for DIR-601.
https://community.phonebooth.com/communi...-150-phonebooth
My PAP2T was working perfect (both inbound and outbound) on voxalot with configuration suggested here: http://forum.voxalot.com/voxalot-general...alkthrough.html
However voxalot is gone now and I switched to PBXes month ago and I made inbound and outbound work without any changes to my router or PAP2T. Only last week I tweaked PAP2T and inbound stopped working.
Any other suggestion. Once my inbound works, I would also like to solve the other problem of calling other extension in my LAN.
I would like to add that my home DID is carried by FUTURE-NINE and I am forwarding it to tjsingh@pbxes.org.
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| Thread: RE: PAP2T/SPA2102: Can't call internal extension and incoming call fails |
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I have two extension 201 and 202 connected to CISCO PAP2T and SPA2102 respectively. I am using D-LINK DIR601 router on my LAN. Both ATAs are on same LAN with IPs 192.168.0.10 and 192.168.0.11. The SIP ports on LINE1 for ATAs are different (5060 and 5062).
I am able to make outbound calls without any problem. Incoming calls ring on extension 201 (PAP2T) however when I pick up the phone connection is not made. For the external party the phone keeps ringing and goes to extension 201 Voice mail.
I have configured my PAP2T as per the following guidelines:
http://faq.voxalot.com/action/view/Setting_up_Linksys_PAP2 (of course the login information is with pbxes not voxalot)
I tired all 3 options for STUN settings with the same result. (one of them with Port forwarding).
At a point incoming calls were working fine and the only issue was that extension 201 was not able to connect to 202 and vice versa (It would ring but would keep rining even if other extension is picked up). In the process of trying to fix that I spoiled the config of PAP2T and incoming stopped working.
Can you help? I think a simple pointer to PAP2T and SPA2012 configuration for pbxes would suffice. I have googled but failed to find one.
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