[Asterisk-Users] Cisco and Asterisk, Weird Stuff
Dave Wolven
dwolven at 123.net
Fri Feb 28 16:34:47 MST 2003
What does the Cisco's config look like.
I'm not sure on the PLAR OPX connection, but would guess a
direct-inword-dial peer would take care of landing the call on the
asterisk box.
I figure something like this may work.
dial-peer voice 1 pots
incoming called-number ..........
direct-inward-dial
digit-strip
prefix [extension to send to asterisk box]
port FXO
!
dial-peer voice 2 voip
destination-pattern [extension to sent to asterisk box]
session-protocol sipv2
session-target ipv4:[ip_of_asterisk_box]
!
dial-peer voice 3 pots
destination-pattern [FXO-extension]
port FXS
Then the normal extension time out would work it magic if the FXS port
was just ringing and then push the caller to voice mail. I wouldn't
worry about the not picking up the port thing, because you will send
them to voice mail anyway.
Dave
On Fri, 2003-02-28 at 15:00, Eric Wieling wrote:
> 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
> ----
>
> 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>
> ----
>
> ;
> ; 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
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