[asterisk-users] adding a second extension
Stephen Reese
rsreese at gmail.com
Thu Oct 23 17:12:20 CDT 2008
I am able to now call the second extension when setup like this so I
believe I'll leave it alone for a while. Basically added the extension
102 to the main incoming line:
exten => 101,1,Dial(SIP/101&SIP/102&SIP/9046260705 at vitel-outbound,30)
exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:)
exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:)
exten => 101,n(lbl_default_0),Hangup()
exten => 101,n(lbl_default_1),Dial(SIP/9046260705 at vitel-outbond,30)
exten => 101,n,Goto(lbl_default_0)
exten => 102,1,Dial(SIP/102,20)
exten => 102,n,Hangup
exten => 102,n,Voicemail(102 at default)
Both extensions can call each other and both extensions ring when the
main line is called... Strange but whatever.
On Thu, Oct 23, 2008 at 1:47 PM, Stephen Reese <rsreese at gmail.com> wrote:
> On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez <jerdguez at gmail.com> wrote:
>> And this phone are connected in a local LAN??
>> Because I see Asterisk receiving a "Bad request" from 68.156.63.118
>> If those phones are not in your local LAN, try with a soft phone first.
>> Could be Zoiper or Xlite.
>> Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
>> sending a "400 Bad request" back to Asterisk.
>>
>
> Both of these phones are on my local lan but the Asterisk server is at
> a colo facility on the internet outside of the local lan. The local
> lan does use NAT/PAT. I see an error "Warning: 399 Bad Request -
> 'Malformed/Missing FROM: field'. Is this a problem?
>
> Thanks
>
> ---
> ns1*CLI>
> <--- SIP read from 68.156.63.118:1082 --->
> INVITE sip:101 at neocipher.net;user=phone SIP/2.0
> Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
> User-Agent: Cisco-CP7912/8.0.1-060412A
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
> Supported: replaces, 100rel
> Expires: 300
> Content-Length: 274
> Content-Type: application/sdp
>
> v=0
> o=102 157742 157742 IN IP4 172.16.2.18
> s=Cisco 7912 SIP Call
> c=IN IP4 68.156.63.118
> t=0 0
> m=audio 16384 RTP/AVP 0 18 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> <------------->
> --- (14 headers 12 lines) ---
> Sending to 68.156.63.118 : 1083 (no NAT)
> Using INVITE request as basis request - 1279503304 at 172.16.2.18
>
> <--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>;tag=as355e0f84
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="ns1.neocipher.net",
> nonce="7c2e1ba9"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '1279503304 at 172.16.2.18' in 32000
> ms (Method: INVITE)
> Found user '102'
>
> <--- SIP read from 64.2.142.116:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
> From: <sip:rsreese at inbound18.vitelity.net>;tag=as401a34d4
> To: <sip:rsreese at inbound18.vitelity.net>
> Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
> CSeq: 3064 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:rsreese at 64.2.142.116>
> Content-Length: 0
>
>
> <------------->
> --- (10 headers 0 lines) ---
>
> <--- SIP read from 64.2.142.116:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
> From: <sip:rsreese at inbound18.vitelity.net>;tag=as401a34d4
> To: <sip:rsreese at inbound18.vitelity.net>;tag=as7a2f92a1
> Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
> CSeq: 3064 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="575628ec"
> Content-Length: 0
>
>
> <------------->
> --- (10 headers 0 lines) ---
> Responding to challenge, registration to domain/host name inbound18.vitelity.net
> REGISTER 13 headers, 0 lines
> Reliably Transmitting (no NAT) to 64.2.142.116:5060:
> REGISTER sip:inbound18.vitelity.net SIP/2.0
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
> From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
> To: <sip:rsreese at inbound18.vitelity.net>
> Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
> CSeq: 3065 REGISTER
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Authorization: Digest username="rsreese", realm="asterisk",
> algorithm=MD5, uri="sip:inbound18.vitelity.net", nonce="575628ec",
> response="b765dbdebba8af18b19707efe651d65d"
> Expires: 120
> Contact: <sip:rsreese at 209.251.157.91>
> Event: registration
> Content-Length: 0
>
>
> ---
>
> <--- SIP read from 68.156.63.118:1082 --->
> ACK sip:101 at neocipher.net;user=phone SIP/2.0
> Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>;tag=as355e0f84
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 1 ACK
> Max-Forwards: 70
> User-Agent: Cisco-CP7912/8.0.1-060412A
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> ns1*CLI>
> <--- SIP read from 68.156.63.118:1082 --->
> INVITE sip:101 at neocipher.net;user=phone SIP/2.0
> Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 2 INVITE
> Max-Forwards: 70
> Contact: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
> User-Agent: Cisco-CP7912/8.0.1-060412A
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
> Supported: replaces, 100rel
> Proxy-Authorization: Digest
> username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:101 at neocipher.net",response="105dfec593cbcfac83380461870c3a07"
> Expires: 300
> Content-Length: 274
> Content-Type: application/sdp
>
> v=0
> o=102 157750 157750 IN IP4 172.16.2.18
> s=Cisco 7912 SIP Call
> c=IN IP4 68.156.63.118
> t=0 0
> m=audio 16384 RTP/AVP 0 18 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> <------------->
> --- (15 headers 12 lines) ---
> Sending to 68.156.63.118 : 1082 (NAT)
> Using INVITE request as basis request - 1279503304 at 172.16.2.18
> Found user '102'
> Found RTP audio format 0
> Found RTP audio format 18
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port 68.156.63.118:16384
> Found audio description format PCMU for ID 0
> Found audio description format G729 for ID 18
> Found audio description format PCMA for ID 8
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
> (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 68.156.63.118:16384
> Looking for 101 in default (domain neocipher.net)
> list_route: hop: <sip:102 at 68.156.63.118:1083;user=phone;transport=udp>
>
> <--- Transmitting (NAT) to 68.156.63.118:1082 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:101 at 209.251.157.91>
> Content-Length: 0
>
>
> <------------>
> -- Executing [101 at default:1] Dial("SIP/102-081e4968",
> "SIP/101&SIP/9046260705 at vitel-outbound|30") in new stack
> Audio is at 209.251.157.91 port 10532
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 68.156.63.118:1038:
> INVITE sip:101 at 68.156.63.118:1039;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
> From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
> To: <sip:101 at 68.156.63.118:1039;transport=udp>
> Contact: <sip:102 at 209.251.157.91>
> Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Thu, 23 Oct 2008 17:39:52 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 289
>
> v=0
> o=root 5235 5235 IN IP4 209.251.157.91
> s=session
> c=IN IP4 209.251.157.91
> t=0 0
> m=audio 10532 RTP/AVP 0 3 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> -- Called 101
> Audio is at 209.251.157.91 port 18610
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x10 (g726aal2) to SDP
> Adding codec 0x20 (adpcm) to SDP
> Adding codec 0x40 (slin) to SDP
> Adding codec 0x80 (lpc10) to SDP
> Adding codec 0x800 (g726) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 64.2.142.17:5060:
> INVITE sip:9046260705 at outbound.vitelity.net SIP/2.0
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
> From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
> To: <sip:9046260705 at outbound.vitelity.net>
> Contact: <sip:rsreese at 209.251.157.91>
> Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Remote-Party-ID: "Stephen\" <sip:102 at 209.251.157.91>;privacy=off;screen=no
> Date: Thu, 23 Oct 2008 17:39:52 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 428
>
> v=0
> o=root 5235 5235 IN IP4 209.251.157.91
> s=session
> c=IN IP4 209.251.157.91
> t=0 0
> m=audio 18610 RTP/AVP 0 3 8 112 5 10 7 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> -- Called 9046260705 at vitel-outbound
> ns1*CLI>
> <--- SIP read from 64.2.142.116:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
> From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
> To: <sip:rsreese at inbound18.vitelity.net>
> Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
> CSeq: 3065 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:rsreese at 64.2.142.116>
> Content-Length: 0
>
>
> <------------->
> --- (10 headers 0 lines) ---
> ns1*CLI>
> <--- SIP read from 64.2.142.116:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
> From: <sip:rsreese at inbound18.vitelity.net>;tag=as751cb0af
> To: <sip:rsreese at inbound18.vitelity.net>;tag=as7a2f92a1
> Call-ID: 44561e1779828d6773b538cd497220d0 at 209.251.157.91
> CSeq: 3065 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Expires: 60
> Contact: <sip:rsreese at 209.251.157.91>;expires=60
> Date: Thu, 23 Oct 2008 17:27:25 GMT
> Content-Length: 0
>
>
> <------------->
> --- (12 headers 0 lines) ---
> Scheduling destruction of SIP dialog
> '44561e1779828d6773b538cd497220d0 at 209.251.157.91' in 32000 ms (Method:
> REGISTER)
> [Oct 23 13:39:52] NOTICE[5264]: chan_sip.c:12682
> handle_response_register: Outbound Registration: Expiry for
> inbound18.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
>
> <--- SIP read from 64.2.142.17:5060 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP
> 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;received=209.251.157.91;rport=5060
> From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
> To: <sip:9046260705 at outbound.vitelity.net>;tag=as0d5d9875
> Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> Transmitting (no NAT) to 64.2.142.17:5060:
> ACK sip:9046260705 at outbound.vitelity.net SIP/2.0
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
> From: "Stephen\" <sip:rsreese at 209.251.157.91>;tag=as0f273e60
> To: <sip:9046260705 at outbound.vitelity.net>;tag=as0d5d9875
> Contact: <sip:rsreese at 209.251.157.91>
> Call-ID: 02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Remote-Party-ID: "Stephen\" <sip:102 at 209.251.157.91>;privacy=off;screen=no
> Content-Length: 0
>
>
> ---
> -- SIP/vitel-outbound-081f9240 is circuit-busy
> ns1*CLI>
> <--- SIP read from 68.156.63.118:1038 --->
> SIP/2.0 400 Bad Request
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
> From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
> To: <sip:101 at 68.156.63.118:1039;transport=udp>
> Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
> Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
> CSeq: 102 INVITE
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> -- Got SIP response 400 "Bad Request" back from 68.156.63.118
> Transmitting (NAT) to 68.156.63.118:1038:
> ACK sip:101 at 68.156.63.118:1039;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
> From: "Stephen\" <sip:102 at 209.251.157.91>;tag=as5b299b78
> To: <sip:101 at 68.156.63.118:1039;transport=udp>
> Contact: <sip:102 at 209.251.157.91>
> Call-ID: 7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> Really destroying SIP dialog
> '02e92ecc7a1c669d51cb1d414167777f at 209.251.157.91' Method: INVITE
> -- SIP/101-08195e68 is circuit-busy
> == Everyone is busy/congested at this time (2:0/2/0)
> -- Executing [101 at default:2] GotoIf("SIP/102-081e4968",
> "0?lbl_default_1:") in new stack
> -- Executing [101 at default:3] GotoIf("SIP/102-081e4968",
> "0?lbl_default_1:") in new stack
> -- Executing [101 at default:4] Hangup("SIP/102-081e4968", "") in new stack
> == Spawn extension (default, 101, 4) exited non-zero on 'SIP/102-081e4968'
> Scheduling destruction of SIP dialog '1279503304 at 172.16.2.18' in 32000
> ms (Method: INVITE)
>
> <--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
> SIP/2.0 603 Declined
> Via: SIP/2.0/UDP
> 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>;tag=as74c3be83
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:101 at 209.251.157.91>
> Content-Length: 0
>
>
> <------------>
> Really destroying SIP dialog
> '7263c1d31ca474285e762cd161cfbeb3 at 209.251.157.91' Method: INVITE
> ns1*CLI>
> <--- SIP read from 68.156.63.118:1082 --->
> ACK sip:101 at neocipher.net;user=phone SIP/2.0
> Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
> From: <sip:102 at neocipher.net;user=phone>;tag=2678814914
> To: <sip:101 at neocipher.net;user=phone>;tag=as74c3be83
> Call-ID: 1279503304 at 172.16.2.18
> CSeq: 2 ACK
> Max-Forwards: 70
> User-Agent: Cisco-CP7912/8.0.1-060412A
> Proxy-Authorization: Digest
> username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:101 at neocipher.net",response="105dfec593cbcfac83380461870c3a07"
> Content-Length: 0
>
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