[Asterisk-Users] Cisco and Asterisk, Weird Stuff
Eric Wieling
eric at fnords.org
Fri Feb 28 13:00:29 MST 2003
I have a Cisco 1750 with 2 FXO ports and 2 FXS ports. I have a
POTS line plugged into FXO port 0 and an Analog phone plugged
into FXS port 0.
I have the FXO port on the Cisco configured as PLAR OPX, which
means that when a call comes into the port the router does NOT
take the port off hook, but DOES initate a VoIP call to the
destination. The destination in this case is an Asterisk box.
Asterisk sees a call coming in for a specific extention and
tries to ring the extention. My analog phone then rings. When
I pick up the phone Asterisk tries a native bridge between the
two points, the Cisco takes the FXO port off hook, and
everything falls apart.
I would have the Cisco just set up as a PLAR OPX connection
directly to the analog phone (that works), but I want the caller
to be able to leave voicemail on the Asterisk server if the
phone is busy or doesn't answer.
Attached are a bunch of debug and sip debug stuff.
--Eric
-------------- next part --------------
NOTICE[8201]: File chan_sip.c, Line 2924 (handle_response): Cancelling timeout 78
sip debug
[0;37;40mSIP Debugging Enabled
*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
Sip read:
INVITE sip:18504844535 at 63.173.166.68:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:68.109.96.14>;tag=56DAFB0-1B13
To: <sip:18504844535 at 63.173.166.68;user=phone>
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 2489756510-365040076-2195233246-4220542755
User-Agent: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731035074
Contact: <sip:68.109.96.14:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 293
v=0
o=CiscoSystemsSIP-GW-UserAgent 8777 4432 IN IP4 68.109.96.14
s=SIP Call
c=IN IP4 68.109.96.14
t=0 0
m=audio 17906 RTP/AVP 4 100 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
18 headers, 13 lines
Interface is eth0
IP Address is 63.173.166.68
Using latest request as basis request
Sending to 68.109.96.14 : 5060
Capabilities: us - 2147483647, them - 1, combined - 1
Looking for 18504844535 in default
Transmitting:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:68.109.96.14>;tag=56DAFB0-1B13
To: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18504844535 at 63.173.166.68;user=phone>
Content-Length: 0
to 68.109.96.14:5060
[1;30;40m == [0;37;40mAccepting call on 'SIP/68.109.96.14:5060' (68.109.96.14)
[1;30;40m -- [0;37;40mExecuting [1;36;40mGoto[0;37;40m("[1;35;40mSIP/68.109.96.14:5060[0;37;40m", "[1;35;40m2113|1[0;37;40m") in new stack
[1;30;40m -- [0;37;40mGoto (default,2113,1)
[1;30;40m -- [0;37;40mExecuting [1;36;40mDial[0;37;40m("[1;35;40mSIP/68.109.96.14:5060[0;37;40m", "[1;35;40mSIP/2113 at 2113|20[0;37;40m") in new stack
Interface is eth0
IP Address is 63.173.166.68
We're at 63.173.166.68 port 10328
Answering with capability 1
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with capability 16
Answering with capability 32
Answering with capability 64
Answering with capability 128
Answering with capability 256
Answering with capability 512
Answering with capability 1024
Answering with capability 2048
Answering with capability 4096
Answering with capability 8192
Answering with capability 16384
Answering with capability 32768
10 headers, 14 lines
XXX Need to handle Retransmitting XXX:
INVITE sip:2113 at 68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
Contact: <sip:681099614 at 63.173.166.68>
To: <sip:2113 at 68.109.96.14>
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 9938 9938 IN IP4 63.173.166.68
s=session
c=IN IP4 63.173.166.68
t=0 0
m=audio 10328 RTP/AVP 4 3 0 8 5 18 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:5 ADPCM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
[1;30;40m -- [0;37;40mCalled 2113 at 2113
WARNING[30730]: File channel.c, Line 1523 (ast_channel_make_compatible): No path to translate from SIP/2113-d862(4) to SIP/68.109.96.14:5060(1)
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 225
v=0
o=CiscoSystemsSIP-GW-UserAgent 6002 9011 IN IP4 68.109.96.14
s=SIP Call
c=IN IP4 68.109.96.14
t=0 0
m=audio 16900 RTP/AVP 4 100
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
12 headers, 10 lines
Capabilities: us - 2147483647, them - 1, combined - 1
DEBUG[8201]: File chan_sip.c, Line 1357 (process_sdp): Oooh, we need to change our formats since our peer supports only 1 and not 4
NOTICE[8201]: File channel.c, Line 1276 (ast_set_read_format): Unable to find a path from 1 to 4
NOTICE[8201]: File channel.c, Line 1247 (ast_set_write_format): Unable to find a path from 4 to 1
We're at 63.173.166.68 port 4632
Answering with capability 1
Transmitting:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:68.109.96.14>;tag=56DAFB0-1B13
To: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18504844535 at 63.173.166.68;user=phone>
Content-Type: application/sdp
Content-Length: 188
v=0
o=root 9938 9938 IN IP4 63.173.166.68
s=session
c=IN IP4 63.173.166.68
t=0 0
m=audio 4632 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
DEBUG[30730]: File rtp.c, Line 644 (ast_rtp_write): Ooh, format changed from 0 to 1
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 776, ms is 127
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 644 (ast_rtp_write): Ooh, format changed from 0 to 1
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 728, ms is 121
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 648, ms is 111
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 848, ms is 136
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
NOTICE[30730]: File rtp.c, Line 167 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible
NOTICE[30730]: File rtp.c, Line 196 (process_rfc3389): Don't know how to handle RFC3389 for receive codec 1
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 1872, ms is 264
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 848, ms is 136
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
DEBUG[30730]: File rtp.c, Line 602 (ast_rtp_raw_write): Difference is 8568, ms is 1101
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:2113 at 68.109.96.14:5060;user=phone>
Content-Type: application/sdp
Content-Length: 225
v=0
o=CiscoSystemsSIP-GW-UserAgent 6002 9011 IN IP4 68.109.96.14
s=SIP Call
c=IN IP4 68.109.96.14
t=0 0
m=audio 16900 RTP/AVP 4 100
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
12 headers, 10 lines
Capabilities: us - 2147483647, them - 1, combined - 1
XXX Need to handle Retransmitting XXX:
ACK sip:2113 at 68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
to 68.109.96.14:5060
[1;30;40m -- [0;37;40mSIP/2113-d862 answered SIP/68.109.96.14:5060
WARNING[30730]: File rtp.c, Line 693 (ast_rtp_write): Not sure about sending format 1 packets
We're at 63.173.166.68 port 4632
Answering with capability 1
Transmitting:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:68.109.96.14>;tag=56DAFB0-1B13
To: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18504844535 at 63.173.166.68;user=phone>
Content-Type: application/sdp
Content-Length: 188
v=0
o=root 9938 9938 IN IP4 63.173.166.68
s=session
c=IN IP4 63.173.166.68
t=0 0
m=audio 4632 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
[1;30;40m -- [0;37;40mAttempting native bridge of SIP/68.109.96.14:5060 and SIP/2113-d862
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:2113 at 68.109.96.14:5060;user=phone>
Content-Type: application/sdp
Content-Length: 225
v=0
o=CiscoSystemsSIP-GW-UserAgent 6002 9011 IN IP4 68.109.96.14
s=SIP Call
c=IN IP4 68.109.96.14
t=0 0
m=audio 16900 RTP/AVP 4 100
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
12 headers, 10 lines
Capabilities: us - 2147483647, them - 1, combined - 1
XXX Need to handle Retransmitting XXX:
ACK sip:2113 at 68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
to 68.109.96.14:5060
NOTICE[29707]: File rtp.c, Line 167 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible
NOTICE[29707]: File rtp.c, Line 196 (process_rfc3389): Don't know how to handle RFC3389 for receive codec 1
Sip read:
ACK sip:18504844535 at 63.173.166.68:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:68.109.96.14>;tag=56DAFB0-1B13
To: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
Date: Tue, 02 Mar 1993 01:17:54 GMT
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
9 headers, 0 lines
We're at 63.173.166.68 port 4632
Answering with capability 1
Transmitting:
INVITE sip:68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 187
v=0
o=root 9938 9938 IN IP4 68.109.96.14
s=session
c=IN IP4 68.109.96.14
t=0 0
m=audio 16900 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
We're at 63.173.166.68 port 10328
Answering with capability 1
Transmitting:
INVITE sip:2113 at 68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 187
v=0
o=root 9938 9938 IN IP4 68.109.96.14
s=session
c=IN IP4 68.109.96.14
t=0 0
m=audio 17906 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
Sip read:
SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 102 INVITE
Content-Length: 0
7 headers, 0 lines
Message is INVITE
Sip read:
SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 103 INVITE
Content-Length: 0
7 headers, 0 lines
[1;30;40m -- [0;37;40mGot SIP response 400 "Bad Request - 'Malformed/Missing Contact field'" back from 68.109.96.14
XXX Need to handle Retransmitting XXX:
ACK sip:2113 at 68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=2ad956c5
From: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
To: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
to 68.109.96.14:5060
DEBUG[8201]: File chan_sip.c, Line 524 (__sip_destroy): Detaching from SIP/2113-d862
DEBUG[30730]: File rtp.c, Line 811 (ast_rtp_bridge): Oooh, something is weird, backing out
We're at 63.173.166.68 port 4632
Answering with capability 1
Transmitting:
INVITE sip:68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 188
v=0
o=root 9938 9938 IN IP4 63.173.166.68
s=session
c=IN IP4 63.173.166.68
t=0 0
m=audio 4632 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 68.109.96.14:5060
DEBUG[30730]: File channel.c, Line 1880 (ast_channel_bridge): Nobody there, continuing...
DEBUG[30730]: File chan_sip.c, Line 660 (sip_hangup): Asked to hangup channel not connected
[1;30;40m == [0;37;40mSpawn extension (default, 2113, 1) exited non-zero on 'SIP/68.109.96.14:5060'
XXX Need to handle Retransmitting XXX:
BYE sip:68.109.96.14 SIP/2.0
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 68.109.96.14:5060
Sip read:
SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
CSeq: 103 INVITE
Content-Length: 0
7 headers, 0 lines
Message is INVITE
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.173.166.68:5060;branch=70e45ea6
From: <sip:18504844535 at 63.173.166.68;user=phone>;tag=0fd27ac5
To: <sip:68.109.96.14>;tag=56DAFB0-1B13
Date: Tue, 02 Mar 1993 01:18:00 GMT
Call-ID: 94691BFE-15C211CC-82DB99DE-FB905F23 at 68.109.96.14
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 103 BYE
9 headers, 0 lines
Interface is eth0
IP Address is 63.173.166.68
DEBUG[8201]: File chan_sip.c, Line 3155 (handle_request): That's odd... Got a response on a call we dont know about.
Sip read:
BYE sip:681099614 at 63.173.166.68:5060 SIP/2.0
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
To: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
Date: Tue, 02 Mar 1993 01:18:00 GMT
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 731035087
CSeq: 101 BYE
Content-Length: 0
11 headers, 0 lines
Interface is eth0
IP Address is 63.173.166.68
Transmitting:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.109.96.14:5060
From: <sip:2113 at 68.109.96.14>;tag=56DB1AC-EA7
To: "asterisk" <sip:681099614 at 63.173.166.68>;tag=398bd35e
Call-ID: 1d987de07cd0d8592742b4d0473e7b3b at 63.173.166.68
CSeq: 101 BYE
User-Agent: Asterisk PBX
Contact: <sip:681099614 at 63.173.166.68>
Content-Length: 0
to 68.109.96.14:56621
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
[0;37;40m*CLI>
-------------- next part --------------
;
; Extentions Configuration for Asterisk
;
[default]
include => extentions
include => long-distance
[extentions]
exten => 18504844535,1,Goto(2113,1)
exten => 2112,1,Dial(SIP/2112 at 2112,20) ; Ring the interface, 20 seconds maximum
exten => 2112,2,Voicemail(u2112) ; If unavailable, send to voicemail w/ unavail announce
exten => 2112,3,Goto(default,2112,1) ; If they press #, return to start
exten => 2112,102,Voicemail(b2112) ; If busy, send to voicemail w/ busy announce
exten => 2112,103,Goto(default,2112,1) ; If they press #, return to start
exten => 2113,1,Dial(SIP/2113 at 2113,20) ; Ring the interface, 20 seconds maximum
exten => 2113,2,Voicemail(2113) ; If unavailable, send to voicemail w/ unavail announce
exten => 2113,3,Hangup
include => long-distance
;
; Create an extension, 2108, for evaulating echo latency.
;
exten => 2108,1,Playback(demo-echotest) ; Let them know what's going on
exten => 2108,2,Echo ; Do the echo test
exten => 2108,3,Playback(demo-echodone) ; Let them know it's over
exten => 2108,4,Hangup
;
; Give voicemail at extension 2109
;
exten => 2109,1,VoicemailMain
exten => 2109,2,Hangup
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
[long-distance]
exten => _91XXXXXXXXXX,1,Dial,SIP/${EXTEN:1}@packet8
exten => _91XXXXXXXXXX,2,Dial,SIP/${EXTEN:1}@iconnect
exten => _91XXXXXXXXXX,3,Congestion
More information about the asterisk-users
mailing list