I have always used gvoive for my outbound calls. With call centric on sp2.. Since my number has ported i have switched my outbound over to callcentric. The phone port dials out fine and everything is great.
But my Vgways(IPPHONES) are now only able to recieve calls from OBI. the get reorder tone if they try to dial out. If i switch it over to gvoice everything works fine again.
here is my incoming call plan
{7960>911:li},{7960>**0:aa},{7960>***:aa2},{7960>(<**1:>(Msp1)):sp1},{7960>(<**2:>(Msp2)):sp2},{7960>(<**8:>(Mli)):li},{7960>(<**9:>(Mpp)):pp},{7960>(Msp1):sp1},{7960>0:ph},{7960>(@@.):},{962>911:li},{7962>**0:aa},{962>***:aa2},{962>(<**1:>(Msp1)):sp1},{962>(<**2:>(Msp2)):sp2},{962>(<**8:>(Mli)):li},{962>(<**9:>(Mpp)):pp},{962>(Msp1):tg1},{962>0:ph},{962>(@@.):}{ph,SP2(7960@10.10.10.202:5062),SP2(962@10.10.10.201:5063}
Here is tg1
sp2,sp1
how wierd..
So i changed out GVOICE for my Voip.ms account as sp1.. Loaak how differently OBI is habdeling the signalling..
with SP1 lead in tg1
ipphone to obi
|Time | 10.10.10.202 |
| | | 10.10.10.200 |
|86587.920| INVITE SDP (g711U g711A g729 telephone-eventRT...pe-101) |SIP From: "xxxxxxx" <sip:7960@10.10.10.200 To:<sip:xxxx976@10.10.10.200
| |(5062) ------------------> (5061) |
|86587.924| 100 Trying| |SIP Status
| |(5062) <------------------ (5061) |
|86587.947| 180 Ringing |SIP Status
| |(5062) <------------------ (5061) |
|86595.503| CANCEL | |SIP Request
| |(5062) ------------------> (5061) |
|86595.506| 487 Request Terminated |SIP Status
| |(5062) <------------------ (5061) |
|86595.507| 200 OK | |SIP Status
| |(5062) <------------------ (5061) |
|86595.610| ACK | |SIP Request
| |(5062) ------------------> (5061) |
Obi to voipms
|Time | 10.10.10.200 |
| | | 64.120.22.242 |
|86587.946| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: <sip:xxx929_obi@chicago.voip.ms To:<sip:xxxxxxx09976@chicago.voip.ms
| |(5060) ------------------> (5060) |
|86587.993| 407 Proxy Authentication Required |SIP Status
| |(5060) <------------------ (5060) |
|86587.996| ACK | |SIP Request
| |(5060) ------------------> (5060) |
|86587.999| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: <sip:xxx929_obi@chicago.voip.ms To:<sip:xxxxxx9976@chicago.voip.ms
| |(5060) ------------------> (5060) |
|86588.058| 100 Trying| |SIP Status
| |(5060) <------------------ (5060) |
|86589.074| 183 Session Progress SDP (g711U g729 telephone...entRTPType-101) |SIP Status
| |(5060) <------------------ (5060) |
|86589.079| RTP (g711U) |RTP Num packets:324 Duration:6.458s SSRC:0x4DE56AF3
| |(16600) <------------------ (18452) |
|86595.509| CANCEL | |SIP Request
| |(5060) ------------------> (5060) |
|86595.543| 487 Request Terminated |SIP Status
| |(5060) <------------------ (5060) |
|86595.546| ACK | |SIP Request
| |(5060) ------------------> (5060) |
|86595.546| 200 OK | |SIP Status
| |(5060) <------------------ (5060) |
Everything is fat and happy as long as tg1 leading line is sp1
now here is the result if i put sp2 as the leading link in tg1.
ip phone to obi (invite seems exactly the same)
|Time | 10.10.10.202 |
| | | 10.10.10.200 |
|87148.124| INVITE SDP (g711U g711A g729 telephone-eventRT...pe-101) |SIP From: "xxxxxxx" <sip:7960@10.10.10.200 To:<sip:xxxxxxx@10.10.10.200
| |(5062) ------------------> (5061) |
|87148.129| 100 Trying| |SIP Status
| |(5062) <------------------ (5061) |
|87148.151| 180 Ringing |SIP Status
| |(5062) <------------------ (5061) |
|87148.349| 486 Busy Here |SIP Status
| |(5062) <------------------ (5061) |
|87148.412| ACK | |SIP Request
| |(5062) ------------------> (5061) |
Here is what pbi sends to callcentric..
|Time | 10.10.10.200 |
| | | 204.11.192.36 |
|87148.150| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: "xxxxx6" <sip:7960@callcentric.com To:<sip:xxxxxx9976@callcentric.com
| |(5061) ------------------> (5060) |
|87148.261| 407 Proxy Authentication Required |SIP Status
| |(5061) <------------------ (5060) |
|87148.263| ACK | |SIP Request
| |(5061) ------------------> (5060) |
|87148.268| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: "5923986" <sip:7960@callcentric.com To:<sip:xxxxx9976@callcentric.com
| |(5061) ------------------> (5060) |
|87148.345| 403 Incorrect Authentication |SIP Status
| |(5061) <------------------ (5060) |
|87148.348| ACK | |SIP Request
| |(5061) ------------------> (5060) |
sip:7960@callcentric.com is not right.. it should be my callcentric username...
If i make a phone call out the phone port its right...
|Time | 10.10.10.200 |
| | | 204.11.192.35 |
|87881.622| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: <sip:xxxxxxxx4976@callcentric.com To:<sip:xxxxx9976@callcentric.com
| |(5061) ------------------> (5060) |
|87881.699| 407 Proxy Authentication Required |SIP Status
| |(5061) <------------------ (5060) |
|87881.702| ACK | |SIP Request
| |(5061) ------------------> (5060) |
|87881.706| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: <sip:xxxxxx4976@callcentric.com To:<sip:xxxxxx9976@callcentric.com
| |(5061) ------------------> (5060) |
|87881.785| 100 Trying| |SIP Status
| |(5061) <------------------ (5060) |
|87881.854| 100 Trying| |SIP Status
| |(5061) <------------------ (5060) |
|87883.560| 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| |(5061) <------------------ (5060) |
|87883.571| RTP (g711U) |RTP Num packets:534 Duration:10.655s SSRC:0xB86A714B
| |(16804) <------------------ (55892) |
|87883.603| RTP (g711U) |RTP Num packets:23 Duration:0.440s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87884.060| 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| |(5061) <------------------ (5060) |
|87884.063| RTP (g711U) |RTP Num packets:49 Duration:0.979s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87885.062| 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| |(5061) <------------------ (5060) |
|87884.063| RTP (g711U) |RTP Num packets:49 Duration:0.979s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87885.063| RTP (g711U) |RTP Num packets:99 Duration:1.979s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87887.060| 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| |(5061) <------------------ (5060) |
|87887.063| RTP (g711U) |RTP Num packets:200 Duration:3.999s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87891.064| 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| |(5061) <------------------ (5060) |
|87891.103| RTP (g711U) |RTP Num packets:150 Duration:2.979s SSRC:0x60685343
| |(16804) ------------------> (55892) |
|87894.094| CANCEL | |SIP Request
| |(5061) ------------------> (5060) |
|87894.165| 200 OK | |SIP Status
| |(5061) <------------------ (5060) |
|87894.171| 487 Request Terminated |SIP Status
| |(5061) <------------------ (5060) |
|87894.174| ACK | |SIP Request
| |(5061) ------------------> (5060) |
i should not that i forced the phone port out sp2 via **2.
In summary why does obi not use the correct info in the invite when sp2 is set as the primary line in tg1?
i am hoping i just have something wrong but im not sure why voip.ms would work fine on sp1.
Quote from: bruss on August 02, 2011, 07:25:18 AM
{7960>(Msp1):sp1}
{962>(Msp1):tg1}
Your PrimaryLine for the 7960 is SP1. Your PrimaryLine for the 962 is TG1, but you're validating it with the SP1 DigitMap.
Quote from: bruss on August 02, 2011, 07:25:18 AM
{7962>**0:aa}
What's a 7962?
Quote from: bruss on August 02, 2011, 07:25:18 AM
{ph,SP2(7960@10.10.10.202:5062),SP2(962@10.10.10.201:5063}
Is SP2 still configured for SIP?
I ripped everything out and started from scratch.. here the Inbound call on sp2 and yes it is sip. Both os my SP are sip now...
{7960>911:li},{7960>**0:aa},{7960>***:aa2},{7960>(<**1:>(Msp1)):sp1},{7960>(<**2:>(Msp2)):sp2},{7960>(<**8:>(Mli)):li},{7960>(<**9:>(Mpp)):pp},{7960>(Msp1):tg1},{7960>0:ph},{7960>(@@.):},{962>911:li},{962>**0:aa},{962>***:aa2},{962>(<**1:>(Msp1)):sp1},{962>(<**2:>(Msp2)):sp2},{962>(<**8:>(Mli)):li},{962>(<**9:>(Mpp)):pp},{962>(Msp1):tg1},{962>0:ph},{962>(@@.):},{ph,SP2(7960@10.10.10.202:5062),SP2(962@10.10.10.201:5063)}
What's a 7962?
I had a fatfinger in there. i fixed it with {962>**0:aa}
change makes no difference if sp2 is leading line in tg1
is this my problem???
{7960>(Msp1):tg1}
{962>(Msp1):tg1}
The digit map on my tg1 is (Msp1)
Quote from: bruss on August 02, 2011, 08:44:34 AM
is this my problem???
{7960>(Msp1):tg1}
{962>(Msp1):tg1}
The digit map on my tg1 is (Msp1)
If nothing references (Mtg1), nothing uses the DigitMap on TG1.
Normally, if your intended PrimaryLine is TG1, it would be:
{7960>(Mtg1):tg1}
{962>(Mtg1):tg1}
These are the rules that would route ordinary outgoing calls.
Quote from: bruss on August 02, 2011, 08:18:08 AM
sip:7960@callcentric.com is not right.. it should be my callcentric username...
Turn off X_SpoofCallerID.
Quote from: RonR on August 02, 2011, 09:06:46 AM
Quote from: bruss on August 02, 2011, 08:18:08 AM
sip:7960@callcentric.com is not right.. it should be my callcentric username...
Turn off X_SpoofCallerID.
this solved my problem... so i guess i just have to live without callerid on my ip phones.