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Outbound call problem

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imark015

Joined
Feb 6, 2014
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Heyall,

I have an Aastra 6730i Phone configured with 3cx.
Internal calls work perfectly.
When i "internally" call to my softphone on iOS, ( wheb i am using cellular network ) It works perfectly. Sound on both sides.

But when i call from the outside number i got from my voip provider, the connection is there, but i dont have any sound. Not at all.
Same when i try to call outside from the aastra. Calling outbound and receiving calls work perfectly at the softphone.

What am i doing wrong>?

Thanks,

Mark
 
imark015 said:
But when i call from the outside number i got from my voip provider, the connection is there, but i dont have any sound. Not at all.
Click to expand...

So...you mean that any incoming call, from your VoIP provider, has no audio, when answered on either device?

Be sure that all the necessary ports are forwarded in your router. There may also be a setting, or some other issue/conflict within your router depending on make/model/features.

Run the 3CX firewall check and be sure that all tests pass.
 
I do have audio from my softphone.
Just not from the Aastra phone.

Internal from my softphone to the aastra and reverse is working perfectly fine.
 
I am running a firewall check right now from the 3cx application.

I'll post the results.
 
Result: Success: Firewall check passed. This configuration is supported
 
imark015 said:
I do have audio from my softphone.
Just not from the Aastra phone.
Click to expand...

So the issue is only when using the Aastra and placing/receiving outside calls then?

If that is the case then I would suspect a Codec issue. The 3CX log, of the call (perhaps verbose mode) may show this.
 
How can this be a codec issue?

Anyway, do you have a suggestion to set it up?

And, i found some provisioning templates here : https://www.888voip.com/support/index.php?/Knowledgebase/Article/View/150/10/3cx-templates-for-aastra-6730i6731i

These are crashing, When i try to set up an extention it says the files are wrong.
Any idea if there are any templates for 3cx 12?
 
btw, i mean to set up a codec. what codec should i use?
 
I'm afraid that I can't help you with auto provisioning, I manually provision all my devices and unfortunately I don't have any Aastra sets. Someone else may be able to help you out with that.

The most common Codecs for good quality calls, where bandwidth is not a concern, are G711a or G711u. With most devices (and the 3CX trunks) you can chose which Codec will be available. When a connection is made a negotiation happens that selects a Codec from the list agreeable to both ends. 3CX will usually transcode if there is no common Codec between endpoints.

Again, check the 3CX logs right after a "failed" Aastra call for any error messages. there may be something else at play here, but I'm now suspecting that it is a configuration issue with your Aastra set.
 
I restored the factory settings of the phone, ill set it up again, and ill post the log.

I hope you can help me with that.

Ill get back :)

Thanks
 
Code: 
08-Feb-2014 21:58:16.946   L:12.2[Line:10000>>0628359823]: Terminating targets, reason:
08-Feb-2014 21:58:16.946   Leg L:12.2[Line:10000>>0628359823] is terminated: Cause: BYE from PBX
08-Feb-2014 21:58:16.946   Terminated from <sip:[email protected]:5060>;tag=as126010d6 to "31337115757"<sip:[email protected]:5060>;tag=fb231c15; reason: LocalBye
08-Feb-2014 21:58:16.946   L:12.2[Line:10000>>0628359823] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=7833992074385725 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-7833992074385725-1---d8754z-;rport
Max-Forwards: 70
Route: <sip:[2001:4D78:500:500:0:0:0:1];lr;r2=on;ftag=fb231c15;did=025.0b15cac7>
Route: <sip:83.143.188.165;r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>;tag=as126010d6
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 3 BYE
Content-Length: 0
08-Feb-2014 21:58:16.946   SendMsg from "31337115757"<sip:[email protected]:5060>;tag=fb231c15 to <sip:[email protected]:5060>;tag=as126010d6
08-Feb-2014 21:58:16.946   [CM503008]: Call(C:12): Call is terminated
08-Feb-2014 21:58:16.946   L:12.2[Line:10000>>0628359823]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
08-Feb-2014 21:58:16.945   Blocking refers for the Call(C:12)
08-Feb-2014 21:58:16.945   Call(C:12) is terminated
08-Feb-2014 21:58:16.944   L:12.1[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="User hangup"
08-Feb-2014 21:58:16.944   Leg L:12.1[Extn] is terminated: Cause: BYE from 62.238.193.252:5060
08-Feb-2014 21:58:16.943   L:12.1[Extn] got Terminated Recv Req BYE from 62.238.193.252:5060 tid=19f6b839337fb95d7 Call-ID=70f3b45b0a1a65c3:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bK19f6b839337fb95d7
Max-Forwards: 70
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25198 BYE
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c09072b5279:39ec0455d0e7929e1ef1be63ce7d639e",uri="sip:[email protected]:5060",response="e6e0612e07136abc5176ccb3cc072468",algorithm=MD5
User-Agent: Aastra 6730i/3.2.2.2077
Content-Length: 0
08-Feb-2014 21:58:16.942   Terminated from "101"<sip:[email protected]:5060>;tag=59948e5f14 to <sip:[email protected]:5060;user=phone>;tag=a60aee03; reason: RemoteBye
08-Feb-2014 21:58:16.942   L:12.1[Extn] Sending: OnSendResp Send 200/BYE from 0.0.0.0:0 tid=19f6b839337fb95d7 Call-ID=70f3b45b0a1a65c3:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bK19f6b839337fb95d7
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25198 BYE
Content-Length: 0
08-Feb-2014 21:58:16.942   SendMsg from <sip:[email protected]:5060;user=phone>;tag=a60aee03 to "101"<sip:[email protected]:5060>;tag=59948e5f14
08-Feb-2014 21:58:11.609   Currently active calls - 1: [12]
08-Feb-2014 21:58:11.012   Session 746 of leg L:12.1[Extn] is confirmed
08-Feb-2014 21:58:11.012   L:12.1[Extn] got Confirmed Recv Req ACK from 62.238.193.252:5060 tid=820ec7337adf9305b Call-ID=70f3b45b0a1a65c3:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bK820ec7337adf9305b
Max-Forwards: 70
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25197 ACK
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c09072b5279:39ec0455d0e7929e1ef1be63ce7d639e",uri="sip:[email protected]:5060;user=phone",response="a20d1eb0fff60ddae33e7a5d9bf879e5",algorithm=MD5
User-Agent: Aastra 6730i/3.2.2.2077
Content-Length: 0
08-Feb-2014 21:58:11.012   ConnCfmd from "101"<sip:[email protected]:5060>;tag=59948e5f14 to <sip:[email protected]:5060;user=phone>;tag=a60aee03
08-Feb-2014 21:58:10.824   [CM503007]: Call(C:12): Line:10000>>0628359823 has joined, contact <sip:[email protected]:5060>
08-Feb-2014 21:58:10.823   [CM503007]: Call(C:12): Extn:101 has joined, contact <sip:[email protected]:5060>
08-Feb-2014 21:58:10.823   ~Target=VoIPline:0628359823@(Ln.10000@Budgetphone)
08-Feb-2014 21:58:10.823   Session 746 of leg L:12.1[Extn] is connected
08-Feb-2014 21:58:10.823   L:12.1[Extn] got Connected.UAS Send 200/INVITE from 0.0.0.0:0 tid=cdf07cbdcc5d49d33 Call-ID=70f3b45b0a1a65c3:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bKcdf07cbdcc5d49d33
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25197 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.33517.465 (33463)
Content-Length: 246

v=0
o=3cxPS 466758926336 297359376385 IN IP4 5.200.12.75
s=3cxPS Audio call
c=IN IP4 5.200.12.75
t=0 0
m=audio 9000 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
08-Feb-2014 21:58:10.823   Connected(UAS) from "101"<sip:[email protected]:5060>;tag=59948e5f14 to <sip:[email protected]:5060;user=phone>;tag=a60aee03
08-Feb-2014 21:58:10.823   L:12.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=cdf07cbdcc5d49d33 Call-ID=70f3b45b0a1a65c3:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bKcdf07cbdcc5d49d33
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25197 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 246

v=0
o=3cxPS 466758926336 297359376385 IN IP4 5.200.12.75
s=3cxPS Audio call
c=IN IP4 5.200.12.75
t=0 0
m=audio 9000 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
08-Feb-2014 21:58:10.822   SendMsg from <sip:[email protected]:5060;user=phone>;tag=a60aee03 to "101"<sip:[email protected]:5060>;tag=59948e5f14
08-Feb-2014 21:58:10.822   [MS010003] C:12.2:Cannot resolve media address '2001:4d78:500:500::1'. Media is rejected
08-Feb-2014 21:58:10.822   MSLogReports
08-Feb-2014 21:58:10.821   ~Target=VoIPline:0628359823@(Ln.10000@Budgetphone)
08-Feb-2014 21:58:10.821   ~Route=Dev:sip:[email protected]:5060
08-Feb-2014 21:58:10.821   ~Target=VoIPline:0628359823@(Ln.10000@Budgetphone)
08-Feb-2014 21:58:10.821   ~Route=Dev:sip:[email protected]:5060
08-Feb-2014 21:58:10.821   L:12.1[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="Call completed elsewhere"
08-Feb-2014 21:58:10.821   ~Route=Dev:sip:[email protected]:5060
08-Feb-2014 21:58:10.821   L:12.2[Line:10000>>0628359823] has joined to L:12.1[Extn]
08-Feb-2014 21:58:10.821   Session 749 of leg L:12.2[Line:10000>>0628359823] is connected
08-Feb-2014 21:58:10.821   L:12.2[Line:10000>>0628359823] got Connected.UAC Recv 200/INVITE from 2001:4d78:500:500::1:5060 tid=d435185f92203357 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2002:5c8:c4b::5c8:c4b]:5060;received=2002:5C8:C4B:0:0:0:5C8:C4B;branch=z9hG4bK-d8754z-d435185f92203357-1---d8754z-;rport=5060
Record-Route: <sip:83.143.188.165;r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Record-Route: <sip:[2001:4D78:500:500:0:0:0:1];r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as126010d6
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Asterisk PBX
Content-Length: 316

v=0
o=root 8641 8641 IN IP6 2001:4d78:500:500::1
s=session
c=IN IP6 2001:4d78:500:500::1
t=0 0
m=audio 27860 RTP/AVP 8 0 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
08-Feb-2014 21:58:10.821   Connected(UAC) from <sip:[email protected]:5060>;tag=as126010d6 to "31337115757"<sip:[email protected]:5060>;tag=fb231c15
08-Feb-2014 21:58:10.820   Remote SDP is set for leg L:12.2[Line:10000>>0628359823]
08-Feb-2014 21:58:10.820   Answer SDP arrived on Leg L:12.2[Line:10000>>0628359823];
OnAnswer Recv 200/INVITE from 2001:4d78:500:500::1:5060 tid=d435185f92203357 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2002:5c8:c4b::5c8:c4b]:5060;received=2002:5C8:C4B:0:0:0:5C8:C4B;branch=z9hG4bK-d8754z-d435185f92203357-1---d8754z-;rport=5060
Record-Route: <sip:83.143.188.165;r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Record-Route: <sip:[2001:4D78:500:500:0:0:0:1];r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as126010d6
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Asterisk PBX
Content-Length: 316

v=0
o=root 8641 8641 IN IP6 2001:4d78:500:500::1
s=session
c=IN IP6 2001:4d78:500:500::1
t=0 0
m=audio 27860 RTP/AVP 8 0 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
08-Feb-2014 21:58:10.820   L:12.2[Line:10000>>0628359823] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=4141be168219df15 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-4141be168219df15-1---d8754z-;rport
Max-Forwards: 70
Route: <sip:[2001:4D78:500:500:0:0:0:1];lr;r2=on;ftag=fb231c15;did=025.0b15cac7>
Route: <sip:83.143.188.165;r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>;tag=as126010d6
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 2 ACK
Proxy-Authorization: Digest username="31337115757",realm="sip1.budgetphone.nl",nonce="52f69b883962864d1ba70d06756dfcb1ce63b15c",uri="sip:[email protected]:5060",response="fdc10d528713ea54953fde7fc08cbd45",algorithm=MD5
Content-Length: 0
08-Feb-2014 21:58:10.820   SendMsg from "31337115757"<sip:[email protected]:5060>;tag=fb231c15 to <sip:[email protected]:5060>;tag=as126010d6
08-Feb-2014 21:57:56.090   L:12.1[Extn] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=cdf07cbdcc5d49d33 Call-ID=70f3b45b0a1a65c3:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bKcdf07cbdcc5d49d33
Contact: <sip:[email protected]:5060;user=phone>
To: <sip:[email protected]:5060;user=phone>;tag=a60aee03
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25197 INVITE
Content-Length: 0
08-Feb-2014 21:57:56.090   SendMsg from <sip:[email protected]:5060;user=phone>;tag=a60aee03 to "101"<sip:[email protected]:5060>;tag=59948e5f14
08-Feb-2014 21:57:56.090   Provisional response arrived for session 749 of Leg L:12.2[Line:10000>>0628359823]
08-Feb-2014 21:57:56.090   L:12.2[Line:10000>>0628359823] got Provisional Recv 180/INVITE from 2001:4d78:500:500::1:5060 tid=d435185f92203357 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [2002:5c8:c4b::5c8:c4b]:5060;received=2002:5C8:C4B:0:0:0:5C8:C4B;branch=z9hG4bK-d8754z-d435185f92203357-1---d8754z-;rport=5060
Record-Route: <sip:83.143.188.165;r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Record-Route: <sip:[2001:4D78:500:500:0:0:0:1];r2=on;lr;ftag=fb231c15;did=025.0b15cac7>
Contact: <sip:[email protected]>
To: <sip:[email protected]:5060>;tag=as126010d6
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
User-Agent: Asterisk PBX
Content-Length: 0
08-Feb-2014 21:57:56.090   Provisional(180) from <sip:[email protected]:5060>;tag=as126010d6 to "31337115757"<sip:[email protected]:5060>;tag=fb231c15
08-Feb-2014 21:57:56.090   [CM505003]: Provider:[Budgetphone] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [GW02] PBX contact: [sip:[email protected]:5060]
08-Feb-2014 21:57:56.090   [CM503002]: Call(C:12): Alerting Line:10000>>0628359823 by contact <sip:[email protected]:5060>
08-Feb-2014 21:57:56.089   Tenant's phonebook has failed to load!
08-Feb-2014 21:57:56.089   Looking for phone number 0628359823 in tenant's 'default' phonebook
08-Feb-2014 21:57:56.088   UacSession 749 has formed leg L:12.2[Line:10000>>0628359823]
08-Feb-2014 21:57:56.088   Answer from <sip:[email protected]:5060>;tag=as126010d6 to "31337115757"<sip:[email protected]:5060>;tag=fb231c15
08-Feb-2014 21:57:54.568   [CM503025]: Call(C:12): Calling T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060] for L:12.1[Extn]
08-Feb-2014 21:57:54.568   Route to L:12.2[Line:10000>>0628359823] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=2e1b2d30d31be250 Call-ID=ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-2e1b2d30d31be250-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "31337115757"<sip:[email protected]:5060>;tag=fb231c15
Call-ID: ODI5M2UyYWE0ZThhZWIxNzA1MjIxYWY1OWNiMDdmZGY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
Content-Length: 331
Remote-Party-ID: "31337115757"<sip:[email protected]:5060>;party=calling

v=0
o=3cxPS 472731615232 321870888961 IN IP4 5.200.12.75
s=3cxPS Audio call
c=IN IP4 5.200.12.75
t=0 0
m=audio 9002 RTP/AVP 0 9 8 3 99 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:110 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
08-Feb-2014 21:57:54.568   Dev(1):[sip:[email protected]:5060 / 31337115757]: PBX contact is resolved to public IP: <sip:[email protected]:5060>
08-Feb-2014 21:57:54.467   Outbound URI is used: sip:83.143.188.165;lr;ftag=as31c52b24;did=cba.ec1c759
08-Feb-2014 21:57:54.467   Added leg L:12.2[Line:10000>>0628359823]
08-Feb-2014 21:57:54.465   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503027]: Call(C:12): From: Extn:101 ("101" <sip:[email protected]:5060>)  to  T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503004]: Call(C:12): Route 3: from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   Line limit check: Current # of calls for line Lc:10000(@Budgetphone[<sip:[email protected]:5060>]) is 0; limit is 4
08-Feb-2014 21:57:54.465   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503027]: Call(C:12): From: Extn:101 ("101" <sip:[email protected]:5060>)  to  T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503004]: Call(C:12): Route 2: from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   Line limit check: Current # of calls for line Lc:10000(@Budgetphone[<sip:[email protected]:5060>]) is 0; limit is 4
08-Feb-2014 21:57:54.465   [Flow] Call(C:12): making call from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503027]: Call(C:12): From: Extn:101 ("101" <sip:[email protected]:5060>)  to  T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   [CM503004]: Call(C:12): Route 1: from L:12.1[Extn] to T:Line:10000>>0628359823@[Dev:sip:[email protected]:5060]
08-Feb-2014 21:57:54.465   Line limit check: Current # of calls for line Lc:10000(@Budgetphone[<sip:[email protected]:5060>]) is 0; limit is 4
08-Feb-2014 21:57:54.465   Call(C:12): Call from Extn:101 to 0628359823 matches outbound rule 'BP'
08-Feb-2014 21:57:54.465   [Flow] Call(C:12): has built target endpoint: Out#:>>Rule{BP}>>0628359823 for call from L:12.1[Extn]
08-Feb-2014 21:57:54.465   [Flow] Target endpoint for 0628359823 is Out#:>>Rule{BP}>>0628359823
08-Feb-2014 21:57:54.465   Selected prefix: 0
08-Feb-2014 21:57:54.465   Looking for outbound rule: dialed = [0628359823], processed: [0628359823]; from-ext:
08-Feb-2014 21:57:54.465   [Flow] Building target endpoint to 0628359823 from "101" <sip:[email protected]:5060>
08-Feb-2014 21:57:54.465   [CM503010]: Call(C:12): Making route(s) from Extn:101 to <sip:[email protected]:5060>
08-Feb-2014 21:57:54.465   Remote SDP is set for leg L:12.1[Extn]
08-Feb-2014 21:57:54.465   OnOffer from "101"<sip:[email protected]:5060>;tag=59948e5f14
08-Feb-2014 21:57:54.465   [CM505001]: Endpoint Extn:101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Aastra 6730i/3.2.2.2077] PBX contact: [sip:[email protected]:5060]

page 2

Code: 
08-Feb-2014 21:57:54.462   Inbound DID: ''; Phonebook Name: ''
08-Feb-2014 21:57:54.462   [CM500002]: Call(C:12): Info on incoming INVITE from Extn:101:
Invite-IN Recv Req INVITE from 62.238.193.252:5060 tid=cdf07cbdcc5d49d33 Call-ID=70f3b45b0a1a65c3:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 62.238.193.252;branch=z9hG4bKcdf07cbdcc5d49d33
Max-Forwards: 70
Route: <sip:5.200.12.75:5060;lr>
Contact: "101"<sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D3193C1>"
To: <sip:[email protected]:5060;user=phone>
From: "101"<sip:[email protected]:5060>;tag=59948e5f14
Call-ID: 70f3b45b0a1a65c3
CSeq: 25197 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c09072b5279:39ec0455d0e7929e1ef1be63ce7d639e",uri="sip:[email protected]:5060;user=phone",response="a20d1eb0fff60ddae33e7a5d9bf879e5",algorithm=MD5
Supported: path, 100rel, replaces
User-Agent: Aastra 6730i/3.2.2.2077
Allow-Events: talk, hold, conference, LocalModeStatus
Content-Length: 318

v=0
o=MxSIP 0 1 IN IP4 62.238.193.252
s=SIP Call
c=IN IP4 62.238.193.252
t=0 0
m=audio 3000 RTP/AVP 8 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
08-Feb-2014 21:57:54.462   [CM503001]: Call(C:12): Incoming call from Extn:101 to <sip:[email protected]:5060>
08-Feb-2014 21:57:54.364   Outbound URI is used: sip:[email protected]:5060;transport=udp
08-Feb-2014 21:57:54.364   IncomingCall: C:12 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
08-Feb-2014 21:57:54.364   Added leg L:C:12.1[No endpoint yet]
08-Feb-2014 21:57:54.363   UasSession 746 started
08-Feb-2014 21:57:54.363   Call from "101"<sip:[email protected]:5060>;tag=59948e5f14 to <sip:[email protected]:5060;user=phone>;tag=a60aee03
 
I got it fixed! Thanks !:)
 
So what did you do?

The only error I can see in the logs is this...

imark015 said:
08-Feb-2014 21:58:10.822 [MS010003] C:12.2:Cannot resolve media address '2001:4d78:500:500::1'. Media is rejected
Click to expand...
 
I installed version 11, and that got my auto provisioning working, and it got my voice working. I dont know what changed exactly, but i got it fixed by downgrading. lol.
 
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