I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests.
I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. I place the call and the cisco phone rings just once (and it shows for a fraction of second the caller id) and then the connections closes as if the called has hunged up. Where is the problem???? Can anyone help me?
here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=alaw allow=ulaw allow=gsm allow=g729
[xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid="Savas Pavlidis" <1239> host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT
[10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729
[419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729
EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten => _3XX,1,Dial(SIP/[EMAIL PROTECTED]) exten => _3XX,n,Congestion
; ; ; exten => 419,1,Dial(SIP/419)
exten => 420,1,Dial(SIP/xlite1) exten => 420,2,Congestion
; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc.
CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 !
This is the SIP transaction
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060
From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5
To: <sip:[EMAIL PROTECTED]>
Date: Mon, 01 Nov 2004 08:12:53 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 2846660713-722735577-3128754082-2282147162
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:[EMAIL PROTECTED]>;party=calling;screen=no;privacy=off
Timestamp: 1099296773
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 235
v=0 o=CiscoSystemsSIP-GW-UserAgent 1213 9211 IN IP4 10.1.1.1 s=SIP Call c=IN IP4 10.1.1.1 t=0 0 m=audio 18644 RTP/AVP 8 101 c=IN IP4 10.1.1.1 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
20 headers, 11 lines
Using latest request as basis request
Sending to 10.1.1.1 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.1:18644
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found no matching peer or user for '10.1.1.1:56528'
Looking for 419 in default
list_route: hop: <sip:[EMAIL PROTECTED]:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.1:5060
From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5
To: <sip:[EMAIL PROTECTED]>;tag=as4996755e
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 10.1.1.1:5060 We're at 10.1.1.250 port 16084 Answering with preferred capability 0x100(G729A) Answering with non-codec capability 0x1(G723) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84 To: <sip:[EMAIL PROTECTED]:5060;user=phone> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 01 Nov 2004 08:22:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214
v=0 o=root 28878 28878 IN IP4 10.1.1.250 s=session c=IN IP4 10.1.1.250 t=0 0 m=audio 16084 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.1.18:5060
Sip read: CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5 To: <sip:[EMAIL PROTECTED]> Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: [EMAIL PROTECTED] CSeq: 101 CANCEL Max-Forwards: 6 Timestamp: 1099296773 Content-Length: 0
10 headers, 0 lines Sending to 10.1.1.1 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5 To: <sip:[EMAIL PROTECTED]>;tag=as4996755e Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 10.1.1.1:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5 To: <sip:[EMAIL PROTECTED]>;tag=as4996755e Call-ID: [EMAIL PROTECTED] CSeq: 101 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 10.1.1.1:5060 Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84 To: <sip:[EMAIL PROTECTED]:5060;user=phone> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0
(no NAT) to 10.1.1.18:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:[EMAIL PROTECTED]>;tag=8BF5F286-8F5 To: <sip:[EMAIL PROTECTED]>;tag=as4996755e Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK
9 headers, 0 lines Destroying call '[EMAIL PROTECTED]'
Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84 To: <sip:[EMAIL PROTECTED]:5060;user=phone> Call-ID: [EMAIL PROTECTED] Date: Mon, 01 Nov 2004 08:12:52 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: [EMAIL PROTECTED]
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: [EMAIL PROTECTED]
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 CANCEL
Server: CSCO/6
Content-Length: 0
9 headers, 0 lines
Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Call-ID: [EMAIL PROTECTED]
Date: Mon, 01 Nov 2004 08:12:52 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
10 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f
From: "319" <sip:[EMAIL PROTECTED]>;tag=as339f0f84
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.1.1.18:5060 Destroying call '[EMAIL PROTECTED]' sip no debug SIP Debugging Disabled
begin:vcard fn:Savas Pavlidis n:Pavlidis;Savas email;internet:[EMAIL PROTECTED] tel;work:+30 2310 573300 tel;fax:+30 2310 752280 x-mozilla-html:FALSE version:2.1 end:vcard
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