-------------------------------sip.conf [general] context=incoming_acanac ; Default context for incoming calls insecure=very recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=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 ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to "no") ;tos=184 ; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=30 ; Max length of incoming registration we allow defaultexpirey=180 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm ; Allow codecs in order of preference ;Been playing with g729, won�t work if disbabled, says, �No supported codecs� ;Otherwise it says, Can�t translate to ZAP. ;allow=g729 ; Allow codecs in order of preference ;allow=ilbc ; Note: codec order is respected only in [general] ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be > rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;progressinband=no ; If we should generate in-band ringing always useragent=Asterisk ; Allows you to change the user agent string canreinvite=yes echocancel=yes echocanelwhenbridge=yes nat=yes ; NAT settings externip=**.***.***.*** localnet=192.168.1.130/255.255.255.0 canreinvite=no [acanac] type=friend username=123456789 fromuser=123456789 secret=******** host=myhost.ca fromdomain=myhost.ca qualify=no dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm ; Allow codecs in order of preference ;allow=g729 ; Allow codecs in order of preference insecure=very [acanac_out] type=peer username=7806283922 secret=********** host=myhost.ca qualify=no dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm ; Allow codecs in order of preference ;allow=g729 ; Allow codecs in order of preference insecure=very ----------------------------------extensions.conf [globals] DYNAMIC_FEATURES=>automon [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten => s,1,Answer() exten => s,2,Dial(Zap/1,10,r) ;exten => s,2,Echo() [clone] ; incoming calls from the FXO port are directed to this context from zapata.conf exten => s,1,Answer() exten => s,2,Dial(Zap/1,10,r) [internal] exten => 4,1,Answer() exten => 4,2,Dial(Zap/6) exten => 5,1,VoiceMail(1234@default) exten => 6,1,VoiceMailMain(1234@default) exten => 7,1,Monitor(wav,/var/spool/asterisk/monitor/recording,m) exten => 8,1,Answer() exten => 8,2,MusicOnHold() exten => 9,1,Answer() exten => 9,2,Dial(Zap/5) [iaxy] exten => 9,1,Answer() exten => 9,2,Dial(Zap/1,10,r) ;exten => 9,1,Dial(Zap/5) [incoming_acanac] exten => phone,1,NoOp(--- ${CALLERID} calling on Acanac (${EXTEN}) ---) ;exten => phone,2,Dial(SIP/ZAP/1,20) exten => phone,2,Dial(Zap/1,10,r) exten => phone,3,Answer exten => phone,4,Wait,1 exten => phone,5,Voicemail(u1234) exten => phone,6,HangUp [outgoing_acanac] exten => _9.,1,Dial(SIP/${EXTEN:2}@acanac,60) exten => _9.,2,Congestion -------------------------------zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] echocancel=yes echotraining=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 ; Produce a dial-tone when fxs pickup. immediate=yes [channels] context=outgoing_acanac ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; Use FXO signalling for an FXS channel channel => 1 ; Telephone attached to port 1 context=incoming ; Incoming calls go to [incoming] in extensions.conf signalling=fxs_ks ; Use FXS signalling for an FXO channel channel => 5 ; PSTN attached to port 2 context=clone ; Incoming calls go to [incoming] in extensions.conf signalling=fxs_ks ; Use FXS signalling for an FXO channel channel => 6 ; PSTN attached to port 2 -------------------------SIP debug with G729 DISABLED 1 active SIP channel *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message **.***.***.** 123456789 3752b4801cb 00117/00000 unkn No 1 active SIP channel *CLI> <-- SIP read from **.***.***.**:5060: INVITE sip:phone@myrouter SIP/2.0 Via: SIP/2.0/UDP **.***.***.**:5060;branch=z9hG4bK6ba76221;rport From: "123456789" ;tag=as766bc73d To: Contact: Call-ID: 4110cf1073d82288691e34b1723ce464@**.***.***.** CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 04 Mar 2006 02:12:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 1731 1731 IN IP4 **.***.***.** s=session c=IN IP4 **.***.***.** t=0 0 m=audio 14856 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 10 lines)--- Using INVITE request as basis request - 4110cf1073d82288691e34b1723ce464@**.***.***.** Sending to **.***.***.** : 5060 (NAT) Found peer 'acanac_out' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port **.***.***.**:14856 Found description format G729 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 3 17:10:18 NOTICE[20044]: chan_sip.c:3588 process_sdp: No compatible codecs! Transmitting (NAT) to **.***.***.**:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP **.***.***.**:5060;branch=z9hG4bK6ba76221;received=**.***.***.**;rport=5060 From: "123456789" ;tag=as766bc73d To: ;tag=as46bd1a1f Call-ID: 4110cf1073d82288691e34b1723ce464@**.***.***.** CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Destroying call '4110cf1073d82288691e34b1723ce464@**.***.***.**' <-- SIP read from **.***.***.**:5060: ACK sip:phone@myrouter SIP/2.0 Via: SIP/2.0/UDP **.***.***.**:5060;branch=z9hG4bK6ba76221;rport From: "123456789" ;tag=as766bc73d To: ;tag=as46bd1a1f Contact: Call-ID: 4110cf1073d82288691e34b1723ce464@**.***.***.** CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --------------------------------------SIP debug with G729 enabled! <-- SIP read from myvoipprovider.ca:5060: INVITE sip:phone@myrouter SIP/2.0 Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK0d45414f;rport From: "123456789" ;tag=as2c79b5ec To: Contact: Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 04 Mar 2006 02:19:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 1731 1731 IN IP4 myvoipprovider.ca s=session c=IN IP4 myvoipprovider.ca t=0 0 m=audio 29888 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 10 lines)--- Using INVITE request as basis request - 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca Sending to myvoipprovider.ca : 5060 (NAT) Found peer 'acanac_out' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port myvoipprovider.ca:29888 Found description format G729 Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for phone in incoming_acanac (domain myrouter) list_route: hop: Transmitting (NAT) to myvoipprovider.ca:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK0d45414f;received=myvoipprovider.ca;rport=5060 From: "123456789" ;tag=as2c79b5ec To: Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing NoOp("SIP/7806283922-db29", "--- "123456789" <123456789> calling on Acanac (phone) ---") in new stack -- Executing Dial("SIP/7806283922-db29", "Zap/1|10|r") in new stack Mar 3 17:17:07 WARNING[20136]: channel.c:2494 ast_request: No translator path exists for channel type Zap (native 68) to 256 Mar 3 17:17:07 NOTICE[20136]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Answer("SIP/7806283922-db29", "") in new stack We're at myrouter port 16874 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to myvoipprovider.ca:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK0d45414f;received=myvoipprovider.ca;rport=5060 From: "123456789" ;tag=as2c79b5ec To: ;tag=as3490dd4d Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 20033 20033 IN IP4 myrouter s=session c=IN IP4 myrouter t=0 0 m=audio 16874 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/7806283922-db29", "1") in new stack <-- SIP read from myvoipprovider.ca:5060: ACK sip:phone@myrouter SIP/2.0 Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK1704c8d7;rport From: "123456789" ;tag=as2c79b5ec To: ;tag=as3490dd4d Contact: Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- (9 headers 0 lines)--- Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message myvoipprovider.ca 7806283922 2d862500539 00101/00102 g729 No Rx: ACK 1 active SIP channel *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message myvoipprovider.ca 7806283922 2d862500539 00101/00102 g729 No Rx: ACK 1 active SIP channel -- Executing VoiceMail("SIP/7806283922-db29", "u1234") in new stack Mar 3 17:17:08 WARNING[20136]: channel.c:2313 set_format: Unable to find a codec translation path from g729 to gsm Mar 3 17:17:08 WARNING[20136]: file.c:820 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/1234/unavail (format g729): No such file or directory Mar 3 17:17:08 WARNING[20136]: channel.c:2313 set_format: Unable to find a codec translation path from g729 to gsm Mar 3 17:17:08 WARNING[20136]: file.c:820 ast_streamfile: Unable to open vm-intro (format g729): No such file or directory == Spawn extension (incoming_acanac, phone, 5) exited non-zero on 'SIP/7806283922-db29' set_destination: Parsing for address/port to send to set_destination: set destination to myvoipprovider.ca, port 5060 Reliably Transmitting (NAT) to myvoipprovider.ca:5060: BYE sip:123456789@myvoipprovider.ca SIP/2.0 Via: SIP/2.0/UDP myrouter:5060;branch=z9hG4bK02a45ef3;rport From: ;tag=as3490dd4d To: "123456789" ;tag=as2c79b5ec Contact: Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 BYE User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from myvoipprovider.ca:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP myrouter:5060;branch=z9hG4bK02a45ef3;received=myrouter;rport=5060 From: ;tag=as3490dd4d To: "123456789" ;tag=as2c79b5ec Call-ID: 2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause:: Normal Clearing --- (11 headers 0 lines)--- Destroying call '2d862500539574a555872b7b3fae0e4a@myvoipprovider.ca' <-- SIP read from myvoipprovider.ca:5060: OPTIONS sip:phone@myrouter SIP/2.0 Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK5d1ae956 From: "asterisk" ;tag=as2c401340 To: Contact: Call-ID: 6bec3dbe41736939012177831e0505b0@myvoipprovider.ca CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sat, 04 Mar 2006 02:19:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- (11 headers 0 lines)--- Looking for phone in incoming_acanac (domain myrouter) Transmitting (NAT) to myvoipprovider.ca:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP myvoipprovider.ca:5060;branch=z9hG4bK5d1ae956;received=myvoipprovider.ca From: "asterisk" ;tag=as2c401340 To: ;tag=as7d95e89b Call-ID: 6bec3dbe41736939012177831e0505b0@myvoipprovider.ca CSeq: 102 OPTIONS User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Accept: application/sdp Content-Length: 0