Switched from GV to PhonePower - intermittent "not configured"
CheapSk86:
Keep us posted. It sounds like the firmware may be helping. If this is true and the service is starting to look good, the PP plan may be something for me to look at.
I have been trying to compare what you get for cost. It looks like PhonePower has every bit of service Anveo has, or more, for a better cost. Can you guys who have chosen it comment to that? Aside from these initial hiccups?
And are you all on 100's or do any of you have 202s?
jlkjslksdjflksdjf:
I have made the switch to PhonePower as well... I also have intermittent not configured issues.
I have enabled syslog on the Obi100; immediately following a reboot the first REGISTER messages returns a 401 Unauthorized response from PhonePower yet my calls go through... wait a couple minutes or hours and the phone no longer works.
The current solution? Reboot Obi100. LAME!
Code:
Syslogd is listening to port# 514:
<145> Reboot is scheduled in 1 seconds
<0> Reboot checking.....<0> Final Cleanup before reboot....
<144> Goodbye! Reboot Now. (reason: 4)
<6> ==== Networking is ready ====
<0> IP Address= 192.168.200.XXX <0> Gateway = 192.168.200.1 <0> Netmask = 255.255.255.0 <3> SYSTEM REBOOTED (Reason: 4, lifecycle: 1377)
<173> ZT: CustomID 1
<0> SLIC_init ...<0> Reset SLIC...<150> Setup Provisioning for system start! 1500
<0> SLIC & DAA is initialized<6> Start Main Service Now<7> Voice Main
<7> [SLIC] DAA command with wrong id: 0, 2, 0, 0, 0, 0<7> [CPT] --- FXS s/w tone generator (ringback) ---
<7> BASESSL:load cert:5
<7> BASESSL:Load certificate ok
<7> REG:Create 0
<7> [SLIC]:Slic#0 ON HOOK
<7> sendto d0400806:5060(597)
REGISTER sip:sip.phonepower.com:5060 SIP/2.0
Call-ID: 49250263@192.168.200.XXX
Content-Length: 0
CSeq: 15767 REGISTER
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
Max-Forwards: 70
To: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>
Via: SIP/2.0/UDP 192.168.200.XXX:5060;branch=z9hG4bK-7862a1cf;rport
User-Agent: OBIHAI/OBi100-1.3.0.2824
Contact: "Me" <sip:XXXXXXXXXX@192.168.200.XXX:5060>;expires=14400;+sip.instance="<urn:uuid:00000000-0000-0000-0000-9cadef10fe5f>"
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE
Supported: replaces
<7> RxFrom:d0400806:5060
SIP/2.0 401 Unauthorized
Call-ID: 49250263@192.168.200.XXX
CSeq: 15767 REGISTER
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
To: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=192.168.22.50+1+6e571c+4390576c
Via: SIP/2.0/UDP 192.168.200.XXX:5060;received=98.150.XXX.XXX;branch=z9hG4bK-7862a1cf;rport=5060;alias
WWW-Authenticate: Digest realm="sip.phonepower.com",nonce="247ae489b0c6",stale=false,algorithm=MD5,qop="auth"
Server: DC-SIP/2.0
Organization: Phone Power
Content-Length: 0
Also here is a snippet of what the logs show when you CANNOT MAKE OR RECEIVE INCOMING CALLS through PhonePower. I get the SIP/2.0 403 Forbidden error. Again the solution is reboot Obi100. LAME!
Code:
<7> [SLIC]:Slic#0 OFF HOOK
<7> [CPT] --- FXS h/w tone generator (stutter)---
<7> [DSP]: ---- H/W DTMF ON (level:1) : 8 @ 273580 ms----<7> [DSP]: ---- H/W DTMF OFF @ 273670 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 273800 ms----<7> [DSP]: ---- H/W DTMF OFF @ 273870 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 274000 ms----<7> [DSP]: ---- H/W DTMF OFF @ 274050 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 7 @ 274180 ms----<7> [DSP]: ---- H/W DTMF OFF @ 274250 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 6 @ 274380 ms----<7> [DSP]: ---- H/W DTMF OFF @ 274450 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 7 @ 274590 ms----<7> [DSP]: ---- H/W DTMF OFF @ 274660 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 274790 ms----<7> [DSP]: ---- H/W DTMF OFF @ 274860 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 0 @ 274990 ms----<7> [DSP]: ---- H/W DTMF OFF @ 275060 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 5 @ 275190 ms----<7> [DSP]: ---- H/W DTMF OFF @ 275260 ms ----
<7> [DSP]: ---- H/W DTMF ON (level:1) : 1 @ 275390 ms----<7> [DSP]: ---- H/W DTMF OFF @ 275460 ms ----
<7> sendto d0400806:5060(962)
INVITE sip:18557671051@sip.phonepower.com:5060 SIP/2.0
Call-ID: 5625b6d4@192.168.200.125
Content-Length: 311
CSeq: 8001 INVITE
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
Max-Forwards: 70
To: <sip:18557671051@sip.phonepower.com>
Via: SIP/2.0/UDP 192.168.200.125:5060;branch=z9hG4bK-308f4669;rport
User-Agent: OBIHAI/OBi100-1.3.0.2824
Contact: "Me" <sip:XXXXXXXXXX@98.150.XXX.XXX:1045>
Expires: 60
Supported: replaces
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE
Remote-Party-ID: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;party=calling;privacy=off
Content-Type: application/sdp
v=0
o=- 28383 1 IN IP4 98.150.XXX.XXX
s=-
c=IN IP4 98.150.XXX.XXX
t=0 0
m=audio 16604 RTP/AVP 0 8 18 104 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian
<7> RxFrom:d0400806:5060
SIP/2.0 100 Trying
Call-ID: 5625b6d4@192.168.200.125
CSeq: 8001 INVITE
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
To: <sip:18557671051@sip.phonepower.com>
Via: SIP/2.0/UDP 192.168.200.125:5060;received=98.150.XXX.XXX;branch=z9hG4bK-308f4669;rport=5060
<7> RxFrom:d0400806:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.200.125:5060;received=98.150.XXX.XXX;branch=z9hG4bK-308f4669;rport=5060
Call-ID: 5625b6d4@192.168.200.125
CSeq: 8001 INVITE
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
To: <sip:18557671051@sip.phonepower.com>;tag=aprqngfrt-88757g300o724
<7> sendto d0400806:5060(397)
ACK sip:18557671051@sip.phonepower.com:5060 SIP/2.0
Call-ID: 5625b6d4@192.168.200.125
Content-Length: 0
CSeq: 8001 ACK
From: "Me" <sip:XXXXXXXXXX@sip.phonepower.com>;tag=SP14bae5f535468a2d9
Max-Forwards: 70
To: <sip:18557671051@sip.phonepower.com>;tag=aprqngfrt-88757g300o724
Via: SIP/2.0/UDP 192.168.200.125:5060;branch=z9hG4bK-308f4669;rport
User-Agent: OBIHAI/OBi100-1.3.0.2824
<7> RTP:Del Channel
<7> [CPT] --- FXS h/w tone generator (ringback)---
<7> [CPT] --- FXS s/w tone generator (sit_1) ---
<7> [CPT] --- FXS s/w tone generator (sit_1) ---
<7> [SLIC]:Slic#0 ONHOOK
MsCint3:
Quote from: MsCint3 on April 26, 2014, 07:09:23 pm
I, too, had these same problems. First off, I chose Phonepower voip. I was knocked off of the system twice. In both cases I had noticed that my config info was missing. The 1st time i manually replaced the info and went to the phone power site to sync the devices. The next morning no service. The second time I configured through Obitalk approved service providers section. I tabbed on the phone power followed the promps and placed phonepower in the sp1 and left googlevoice in the sp2 section. After the configs were done, I returned to the phonepower site to sync the devices. Earlier I had tried to remove GV to replace with CircleNet voip in sp2. It messed up everything so I reconfig to GV. Now, I have been on for 2 days with no probs.
I hope that this advice works for you as well it has for me.
Good Luck!!
I have been on for 1 week now without any problems.
CheapSk86:
MsCint: are you using a 100 as well or a different model Obi?
MsCint3:
Obi 100
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