Hi there I have asterisk 10.11.1 which seems to have problem negotiating codec.
Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, h263p. I have tried similar combination of codecs and SIP phone but when making a video call, it report "Peer doesn't provide video". It seems Asterisk is failing to set capability correct. Both codecs are enabled on the SIP Phones --- (12 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video Here is a sip show peer output and log when making calls. localhost*CLI> sip show peer 1003 * Name : 1003 Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : video-users Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 1003@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "device" <1003> MaxCallBR : 384 kbps Expire : 3605 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.10.10.129:48464 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1003 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (8 ms) Useragent : X-Lite release 4.5.2 stamp 70142 Reg. Contact : sip:1003@10.10.10.129:48464;rinstance=cf0c3558f05c89dc Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI> sip show peer 1004 * Name : 1004 Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : video-users Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 1004@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "device" <1004> MaxCallBR : 384 kbps Expire : 893 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.10.10.107:21769 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1004 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (2 ms) Useragent : Grandstream GXV3175v2 1.0.1.19 Reg. Contact : sip:1004@10.10.10.107:21769 Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI> <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.10.10.129:48464 ---> INVITE sip:1004@10.10.10.105 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1003@10.10.10.129:48464> To: <sip:1004@10.10.10.105> From: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5.2 stamp 70142 Authorization: Digest username="1003",realm="10.10.10.105",nonce="05e8af6e",uri="sip:1004@10.10.10.105",response="20e63a04aa86d6ec1d1e045c05159b39",algorithm=MD5 Content-Length: 418 v=0 o=- 13015615910543193 1 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 49490 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli a=sendrecv <-------------> --- (14 headers 16 lines) --- Sending to 10.10.10.129:48464 (NAT) Using INVITE request as basis request - MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA Found peer '1003' for '1003' from 10.10.10.129:48464 == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found RTP video format 115 Found RTP video format 34 Found video description format H263-1998 for ID 115 Found video description format H263 for ID 34 Capabilities: us - (alaw|h263p), peer - audio=(ulaw|alaw)/video=(h263|h263p)/text=(nothing), combined - (alaw|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer video RTP is at port 10.10.10.129:49490 Looking for 1004 in video-users (domain 10.10.10.105) list_route: hop: <sip:1003@10.10.10.129:48464> <--- Transmitting (NAT) to 10.10.10.129:48464 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;received=10.10.10.129;rport=48464 From: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c To: <sip:1004@10.10.10.105> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 INVITE Server: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1004@10.10.10.105:5060> Content-Length: 0 <------------> -- Executing [1004@video-users:1] Answer("SIP/1003-00000020", "") in new stack Audio is at 13410 Video is at 10.10.10.105:13834 Adding codec 100004 (alaw) to SDP Adding video codec 200003 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;received=10.10.10.129;rport=48464 From: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c To: <sip:1004@10.10.10.105>;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 INVITE Server: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1004@10.10.10.105:5060> Content-Type: application/sdp Content-Length: 315 v=0 o=root 1557854096 1557854096 IN IP4 10.10.10.105 s=Asterisk PBX 10.11.1 c=IN IP4 10.10.10.105 b=CT:384 t=0 0 m=audio 13410 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 13834 RTP/AVP 115 a=rtpmap:115 h263-1998/90000 a=sendrecv <------------> <--- SIP read from UDP:10.10.10.129:48464 ---> ACK sip:1004@10.10.10.105:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-2ee063296946cc3e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1003@10.10.10.129:48464> To: <sip:1004@10.10.10.105>;tag=as24914503 From: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 ACK User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Executing [1004@video-users:2] Dial("SIP/1003-00000020", "SIP/1004") in new stack == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Called SIP/1004 -- SIP/1004-00000021 is ringing -- SIP/1004-00000021 answered SIP/1003-00000020 -- Remotely bridging SIP/1003-00000020 and SIP/1004-00000021 set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Audio is at 13410 Video is at 10.10.10.107:57822 Adding codec 100004 (alaw) to SDP Adding video codec 200003 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.10.129:48464: INVITE sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 INVITE User-Agent: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 315 v=0 o=root 1557854096 1557854097 IN IP4 10.10.10.107 s=Asterisk PBX 10.11.1 c=IN IP4 10.10.10.107 b=CT:384 t=0 0 m=audio 53104 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 57822 RTP/AVP 115 a=rtpmap:115 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 Contact: <sip:1003@10.10.10.129:48464> To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c From: <sip:1004@10.10.10.105>;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 234 v=0 o=- 13015615910543193 2 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 0 RTP/AVP 115 <-------------> --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Transmitting (NAT) to 10.10.10.129:48464: ACK sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK24550383;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 ACK User-Agent: Asterisk PBX 10.11.1 Content-Length: 0 --- set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Audio is at 13410 Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.10.129:48464: INVITE sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 103 INVITE User-Agent: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 236 v=0 o=root 1557854096 1557854098 IN IP4 10.10.10.107 s=Asterisk PBX 10.11.1 c=IN IP4 10.10.10.107 t=0 0 m=audio 53104 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #1 (NAT) to 10.10.10.129:48464: INVITE sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 103 INVITE User-Agent: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 236 v=0 o=root 1557854096 1557854098 IN IP4 10.10.10.107 s=Asterisk PBX 10.11.1 c=IN IP4 10.10.10.107 t=0 0 m=audio 53104 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport=5060 Contact: <sip:1003@10.10.10.129:48464> To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c From: <sip:1004@10.10.10.105>;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 211 v=0 o=- 13015615910543193 3 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Transmitting (NAT) to 10.10.10.129:48464: ACK sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK2f83ceba;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 103 ACK User-Agent: Asterisk PBX 10.11.1 Content-Length: 0 --- set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Audio is at 13410 Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.10.129:48464: INVITE sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK062304c6;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 104 INVITE User-Agent: Asterisk PBX 10.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 236 v=0 o=root 1557854096 1557854099 IN IP4 10.10.10.107 s=Asterisk PBX 10.11.1 c=IN IP4 10.10.10.107 t=0 0 m=audio 53104 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK062304c6;rport=5060 Contact: <sip:1003@10.10.10.129:48464> To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c From: <sip:1004@10.10.10.105>;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 211 v=0 o=- 13015615910543193 4 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video set_destination: Parsing <sip:1003@10.10.10.129:48464> for address/port to send to set_destination: set destination to 10.10.10.129:48464 Transmitting (NAT) to 10.10.10.129:48464: ACK sip:1003@10.10.10.129:48464 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK490ee170;rport Max-Forwards: 70 From: <sip:1004@10.10.10.105>;tag=as24914503 To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c Contact: <sip:1004@10.10.10.105:5060> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 104 ACK User-Agent: Asterisk PBX 10.11.1 Content-Length: 0 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users