My setup:
DID Number points to the server, hears IVR then user enters pin, then dials outgoing VOIP call. The IVR sounds perfect, the call connects externally. The outgoing VOIP call sounds fine the first few seconds, then it gets really choppy, then within a minute, the call drops.
Tried it with differnt recipient and originating phones. The other weird thing is that only the caller gets the choppy quality. The Receiver sounds great the whole time.
Please help! here’s my SIP Debug log:
--- (14 headers 15 lines) ---
Sending to 204.11.192.133:5068 (no NAT)
Sending to 204.11.192.133:5068 (no NAT)
Using INVITE request as basis request - 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
No matching peer for '85264664514' from '204.11.192.133:5068'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.133:13482
Looking for 6001 in incoming (domain 192.168.100.178)
list_route: route/path hop: <sip:85264664514@204.11.192.133:5068>
<--- Transmitting (no NAT) to 204.11.192.133:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.133:5068;branch=z9hG4bK1eaa5176;received=204.11.192.133;rport=5068
From: "85264664514" <sip:85264664514@204.11.192.133:5068>;tag=as4953f72e
To: <sip:6001@192.168.100.178>
Call-ID: 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
CSeq: 102 INVITE
Server: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6001@192.168.100.178:5060>
Content-Length: 0
<------------>
Audio is at 15610
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 204.11.192.133:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.133:5068;branch=z9hG4bK1eaa5176;received=204.11.192.133;rport=5068
From: "85264664514" <sip:85264664514@204.11.192.133:5068>;tag=as4953f72e
To: <sip:6001@192.168.100.178>;tag=as075f73d4
Call-ID: 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
CSeq: 102 INVITE
Server: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6001@192.168.100.178:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 308
v=0
o=root 1398678438 1398678438 IN IP4 192.168.100.178
s=Asterisk PBX 12.1.1
c=IN IP4 192.168.100.178
t=0 0
m=audio 15610 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:204.11.192.133:5068 --->
ACK sip:6001@192.168.100.178:5060 SIP/2.0
Via: SIP/2.0/UDP 204.11.192.133:5068;branch=z9hG4bK00b35751;rport
Max-Forwards: 70
From: "85264664514" <sip:85264664514@204.11.192.133:5068>;tag=as4953f72e
To: <sip:6001@192.168.100.178>;tag=as075f73d4
Contact: <sip:85264664514@204.11.192.133:5068>
Call-ID: 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
CSeq: 102 ACK
User-Agent: Callcentric-b2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '248fd7d86693d593507a9f0428de4c2d@204.11.192.133' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:85264664514@204.11.192.133:5068> for address/port to send to
set_destination: set destination to 204.11.192.133:5068
Reliably Transmitting (no NAT) to 204.11.192.133:5068:
BYE sip:85264664514@204.11.192.133:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK5fa1e56a;rport
Max-Forwards: 70
From: <sip:6001@192.168.100.178>;tag=as075f73d4
To: "85264664514" <sip:85264664514@204.11.192.133:5068>;tag=as4953f72e
Call-ID: 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.1.1
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.133:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK5fa1e56a;received=192.168.100.178;rport=5060
From: <sip:6001@192.168.100.178>;tag=as075f73d4
To: "85264664514" <sip:85264664514@204.11.192.133:5068>;tag=as4953f72e
Call-ID: 248fd7d86693d593507a9f0428de4c2d@204.11.192.133
CSeq: 102 BYE
Server: Callcentric-b2b
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '248fd7d86693d593507a9f0428de4c2d@204.11.192.133' Method: ACK
[Apr 8 17:38:33] NOTICE[9336]: chan_sip.c:15261 sip_reregister: -- Re-registration for 17772189338@callcentric.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.170:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK43a2cbf2
Max-Forwards: 70
From: <sip:17772189338@callcentric.com>;tag=as3fb0c7df
To: <sip:17772189338@callcentric.com>
Call-ID: 413d323128c160465845feaf510d0544@192.168.100.178
CSeq: 3774 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 12.1.1
Authorization: Digest username="17772189338", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="54cb2c62ec38c102c5e8504d550cff94", response="347217cf265491aa0bda40b6a479a37b"
Expires: 120
Contact: <sip:s@192.168.100.178:5060>
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.170:5060 --->
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK43a2cbf2
f: <sip:17772189338@callcentric.com>;tag=as3fb0c7df
t: <sip:17772189338@callcentric.com>
i: 413d323128c160465845feaf510d0544@192.168.100.178
CSeq: 3774 REGISTER
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="f80a8333decf235bd2099f5fd238d0fd", opaque="", stale=TRUE, algorithm=MD5
l: 0
<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name callcentric.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.170:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2efd221f
Max-Forwards: 70
From: <sip:17772189338@callcentric.com>;tag=as3fb0c7df
To: <sip:17772189338@callcentric.com>
Call-ID: 413d323128c160465845feaf510d0544@192.168.100.178
CSeq: 3775 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 12.1.1
Proxy-Authorization: Digest username="17772189338", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="f80a8333decf235bd2099f5fd238d0fd", response="cfbffe0307eaddfbe6feaa853b45d62c"
Expires: 120
Contact: <sip:s@192.168.100.178:5060>
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.170:5060 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2efd221f
f: <sip:17772189338@callcentric.com>;tag=as3fb0c7df
t: <sip:17772189338@callcentric.com>
i: 413d323128c160465845feaf510d0544@192.168.100.178
CSeq: 3775 REGISTER
m: <sip:s@192.168.100.178:5060>;expires=60
l: 0
<------------->
--- (8 headers 0 lines) ---
[Apr 8 17:38:34] NOTICE[9336]: chan_sip.c:24015 handle_response_register: Outbound Registration: Expiry for callcentric.com is 60 sec (Scheduling reregistration in 45 s)
Really destroying SIP dialog '413d323128c160465845feaf510d0544@192.168.100.178' Method: REGISTER
<--- SIP read from UDP:210.3.88.146:1216 --->
<------------->
Reliably Transmitting (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=9817096540e0d850;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK297660fd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as77d548e9
To: <sip:6001@210.3.88.146:1216;rinstance=9817096540e0d850;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 57de7d0b1dd82e6d56983e2410e6a73d@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Tue, 08 Apr 2014 09:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:210.3.88.146:1216 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK297660fd
Contact: <sip:192.168.1.74:51187>
To: <sip:6001@210.3.88.146:1216;rinstance=9817096540e0d850;transport=UDP>;tag=244a2845
From: "asterisk"<sip:asterisk@192.168.100.178>;tag=as77d548e9
Call-ID: 57de7d0b1dd82e6d56983e2410e6a73d@192.168.100.178:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '57de7d0b1dd82e6d56983e2410e6a73d@192.168.100.178:5060' Method: OPTIONS
<--- SIP read from UDP:204.11.192.133:5068 --->
INVITE sip:6001@192.168.100.178 SIP/2.0
Via: SIP/2.0/UDP 204.11.192.133:5068;branch=z9hG4bK2a5916c9;rport
Max-Forwards: 70
From: "15264664514" <sip:15264664514@204.11.192.133:5068>;tag=as4edf5c2d
To: <sip:6001@192.168.100.178>
Contact: <sip:15264664514@204.11.192.133:5068>
Call-ID: 39c992f558e2e05a15b7067e6264356f@204.11.192.133
CSeq: 102 INVITE
User-Agent: Callcentric-b2b
Date: Tue, 08 Apr 2014 09:38:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 323