# Asterisk problems with outgoing calls **URL:** \ **Category:** Asterisk Support\ **Created:** [December 4, 2013, 10:47am UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667 "2013-12-04T10:47:07Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![bialyzs3](https://avatars.discourse-cdn.com/v4/letter/b/b19c9b/32.png) [@bialyzs3](https://community.asterisk.org/u/bialyzs3)\ **Post date:** [December 4, 2013, 10:47am UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667/1 "2013-12-04T10:47:07Z")
Hi, I am newbie in Asterisk and I have a problem with outgoing call’s from my asterisk to my voip provider (freeconet)… Softwear: CentOS 6.4 Asterisk 1.8.24 FreePBX 2.9 X-Lite (soft phone) Topology: My provider (freeconet) - Asterisk (on CentOS) - X-lite I can call on my voip number with succes, unfortunately when I tried to call from X-lite my call is automatically terminated. Statement on ma soft phone: > [@](#): > > Failed to estabilish call Statement in terminal from Asterisk : [quote]==Using SIP RTP TOS bits 184 ==Using SIP RTP CoS mark 5[/quote] Logfiles from my freePBX when i call to voip phone and from voip phone: [quote][Nov 29 08:18:55] VERBOSE[3647] asterisk.c: Asterisk Ready. [Nov 29 08:19:03] VERBOSE[3649] asterisk.c: – Remote UNIX connection [Nov 29 08:19:08] VERBOSE[3672] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 29 08:19:08] VERBOSE[3672] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [przychodzace1@freeconet:1] Set(“SIP/freeconet-in1-00000000”, “TOHDR=”) in new stack [Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [przychodzace1@freeconet:2] GotoIf(“SIP/freeconet-in1-00000000”, “1?wew,133,1”) in new stack [Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Goto (wew,133,1) [Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [133@wew:1] Dial(“SIP/freeconet-in1-00000000”, “SIP/3784433,45,Tt”) in new stack [Nov 29 08:19:08] VERBOSE[3691] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 29 08:19:08] VERBOSE[3691] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 29 08:19:08] VERBOSE[3691] app\_dial.c: – Called SIP/3784433 [Nov 29 08:19:09] VERBOSE[3691] app\_dial.c: – SIP/3784433-00000001 is ringing [Nov 29 08:19:12] VERBOSE[3691] pbx.c: == Spawn extension (wew, 133, 1) exited non-zero on ‘SIP/freeconet-in1-00000000’ [Nov 29 08:21:39] VERBOSE[3672] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 29 08:21:39] VERBOSE[3672] netsock2.c: == Using SIP RTP CoS mark 5[/quote] My sip configuration: [quote] context=default bindport=5060 ``` srvlookup=yes defaultexpiry=60 allowguest=no dtmfmode=rfc2833 nat=yes defaultexpiry=60 localnet=192.168.2.1/255.255.255.0 externip=77.252.253.170 register => login:pass@sip.freeconet.pl/przychodzace1 [freeconet-out] type=peer username=login secret=pass fromdomain=sip.freeconet.pl context=freeconet host=sip.freeconet.pl port=5060 outboundproxy=sip.freeconet.pl outboundproxyport=5060 insecure=no [freeconet-in1] type=peer fromdomain=sip.freeconet.pl port=5060 context=freeconet [/quote] ``` My extensions file: [quote] [defaul] exten =\> \_.,1,Hangup ``` [freeconet] exten =>przychodzace1,1,Set(TOHDR=${SIP_HEADER(To)}) exten =>przychodzace1,2,GotoIf($["${REGEX("223784433" ${TOHDR})}" = "1"]?wew,133,1) [freeconet1] include => wew exten => t,1,Hangup exten => h,1,Hangup exten => _XXX.,1,SIPAddHeader(X-Fid: ${SIPCALLID}) exten => _XXX.,2,Set(CALLERID(num)=48223784433) exten => _XXX.,3,Dial(SIP/${EXTEN}@freeconet-out) [wew] exten => 133,1,Dial(SIP/3784433,45,Tt)[/quote] ``` sip show peers: [quote]Name/username Host Dyn Forcerport ACL Port Status 3784433/3784433 192.168.2.104 D N 5060 Unmonitored freeconet-in1 213.218.116.65 N 5060 Unmonitored freeconet-in2 213.218.116.66 N 5060 Unmonitored freeconet-out/lukaszbiele 213.218.116.66 N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] [/quote] sip show registry: [quote]Host dnsmgr Username Refresh State Reg.Time sip.freeconet.pl:5060 N lukaszbielec 45 Registered Thu, 28 Nov 2013 12:33:59 1 SIP registrations.[/quote] sip set debug peer 3784433 and call from voip phone: > [@](#): > > ``` > <--- SIP read from UDP:192.168.2.104:52610 ---> > INVITE sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;rport > Max-Forwards: 70 > Contact: > To: > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces > User-Agent: X-Lite release 4.5.5 stamp 71236 > Content-Length: 306 > > v=0 > o=- 13030197534578821 1 IN IP4 192.168.2.104 > s=X-Lite 4 release 4.5.5 stamp 71236 > c=IN IP4 192.168.2.104 > t=0 0 > m=audio 57762 RTP/AVP 125 9 8 0 100 101 > a=rtpmap:125 opus/48000/2 > a=fmtp:125 useinbandfec=1 > a=rtpmap:100 speex/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > <-------------> > --- (13 headers 12 lines) --- > Sending to 192.168.2.104:52610 (NAT) > Using INVITE request as basis request - MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > Found peer '3784433' for '3784433' from 192.168.2.104:52610 > > <--- Reliably Transmitting (NAT) to 192.168.2.104:52610 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;received=192.168.2.104;rport=52610 > From: ;tag=18b3b07c > To: ;tag=as4b6ee874 > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 INVITE > Server: FPBX-2.9.0(1.8.24.0) > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6896b6b2" > Content-Length: 0 > > <------------> > Scheduling destruction of SIP dialog 'MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc' in 32000 ms (Method: INVITE) > Retransmitting #1 (NAT) to 192.168.2.104:52610: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;received=192.168.2.104;rport=52610 > From: ;tag=18b3b07c > To: ;tag=as4b6ee874 > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 INVITE > Server: FPBX-2.9.0(1.8.24.0) > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6896b6b2" > Content-Length: 0 > > --- > > <--- SIP read from UDP:192.168.2.104:52610 ---> > INVITE sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;rport > Max-Forwards: 70 > Contact: > To: > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces > User-Agent: X-Lite release 4.5.5 stamp 71236 > Content-Length: 306 > > v=0 > o=- 13030197534578821 1 IN IP4 192.168.2.104 > s=X-Lite 4 release 4.5.5 stamp 71236 > c=IN IP4 192.168.2.104 > t=0 0 > m=audio 57762 RTP/AVP 125 9 8 0 100 101 > a=rtpmap:125 opus/48000/2 > a=fmtp:125 useinbandfec=1 > a=rtpmap:100 speex/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > <-------------> > --- (13 headers 12 lines) --- > Ignoring this INVITE request > > <--- SIP read from UDP:192.168.2.104:52610 ---> > ACK sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;rport > Max-Forwards: 70 > To: ;tag=as4b6ee874 > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 ACK > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > > <--- SIP read from UDP:192.168.2.104:52610 ---> > INVITE sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-37ad243a1a026a1b-1---d8754z-;rport > Max-Forwards: 70 > Contact: > To: > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces > User-Agent: X-Lite release 4.5.5 stamp 71236 > Authorization: Digest username="3784433",realm="asterisk",nonce="6896b6b2",uri="sip:900@192.168.2.111",response="b51c87ea9ae3dc3488bc9a234340b1c7",algorithm=MD5 > Content-Length: 306 > > v=0 > o=- 13030197534578821 1 IN IP4 192.168.2.104 > s=X-Lite 4 release 4.5.5 stamp 71236 > c=IN IP4 192.168.2.104 > t=0 0 > m=audio 57762 RTP/AVP 125 9 8 0 100 101 > a=rtpmap:125 opus/48000/2 > a=fmtp:125 useinbandfec=1 > a=rtpmap:100 speex/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > <-------------> > --- (14 headers 12 lines) --- > Sending to 192.168.2.104:52610 (NAT) > Using INVITE request as basis request - MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > Found peer '3784433' for '3784433' from 192.168.2.104:52610 > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > Found RTP audio format 125 > Found RTP audio format 9 > Found RTP audio format 8 > Found RTP audio format 0 > Found RTP audio format 100 > Found RTP audio format 101 > Found unknown media description format opus for ID 125 > Found audio description format speex for ID 100 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20000100c (ulaw|alaw|speex16|g722) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > Peer audio RTP is at port 192.168.2.104:57762 > Looking for 900 in freeconet1 (domain 192.168.2.111) > > <--- Reliably Transmitting (NAT) to 192.168.2.104:52610 ---> > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-37ad243a1a026a1b-1---d8754z-;received=192.168.2.104;rport=52610 > From: ;tag=18b3b07c > To: ;tag=as4b6ee874 > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 2 INVITE > Server: FPBX-2.9.0(1.8.24.0) > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 > > <------------> > Scheduling destruction of SIP dialog 'MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc' in 32000 ms (Method: INVITE) > > <--- SIP read from UDP:192.168.2.104:52610 ---> > ACK sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-a1f24a611943cb62-1---d8754z-;rport > Max-Forwards: 70 > To: ;tag=as4b6ee874 > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 1 ACK > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > > <--- SIP read from UDP:192.168.2.104:52610 ---> > ACK sip:900@192.168.2.111 SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.104:52610;branch=z9hG4bK-d8754z-37ad243a1a026a1b-1---d8754z-;rport > Max-Forwards: 70 > To: ;tag=as4b6ee874 > From: ;tag=18b3b07c > Call-ID: MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc > CSeq: 2 ACK > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > > <--- SIP read from UDP:192.168.2.104:52610 ---> > > <-------------> > Really destroying SIP dialog 'MzQ4MTg2MTc1Nzc3MGI1NGFmYjUyNzViNDA5NzNlOTc' Method: ACK > > <--- SIP read from UDP:192.168.2.104:52610 ---> > > <-------------> > > ``` Support from my voip provider don’t want help… In connection with this I ask you for help… Do you have solution or any suggestions? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [December 4, 2013, 11:14am UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667/2 "2013-12-04T11:14:56Z")
SIP/2.0 484 Address Incomplete They’ve sent a 3 digit number (900). Your dialplan expects a number of 4 or more digits (. means 1 or more digits). Note that the RTP COS messages are not errors. ---
**Author:** ![bialyzs3](https://avatars.discourse-cdn.com/v4/letter/b/b19c9b/32.png) [@bialyzs3](https://community.asterisk.org/u/bialyzs3)\ **Post date:** [December 4, 2013, 2:26pm UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667/3 "2013-12-04T14:26:56Z")
Unfortunately, still I can’t make outgoing calls… When i call to other number (1001213): Log’s form terminal: [quote]== Using SIP RTP CoS mark 5 – Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000004”, “X-Fid: Zjc0ODFkZjQ1MzkyYTEyMGRlZDBhMThjYTUxZGQ1NjU”) in new stack – Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000004”, “CALLERID(num)=48223784433”) in new stack – Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000004”, “SIP/1001213@freeconet-out”) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 – Called SIP/1001213@freeconet-out – SIP/freeconet-out-00000005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) – Auto fallthrough, channel ‘SIP/3784433-00000004’ status is ‘CONGESTION’ – Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000004”, “”) in new stack == Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000004’[/quote] sip set debug peer 3784433: [quote]\<— SIP read from UDP:192.168.2.103:5060 —\> INVITE [sip:1001213@192.168.2.101](mailto:sip:1001213@192.168.2.101) SIP/2.0 Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;rport Max-Forwards: 70 Contact: sip:3784433@192.168.2.103:5060 To: sip:1001213@192.168.2.101 From: sip:3784433@192.168.2.101;tag=a76b7847 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5.5 stamp 71236 Content-Length: 306 v=0 o=- 13030640618233466 1 IN IP4 192.168.2.103 s=X-Lite 4 release 4.5.5 stamp 71236 c=IN IP4 192.168.2.103 t=0 0 m=audio 50394 RTP/AVP 125 9 8 0 100 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv \<-------------\> — (13 headers 12 lines) — Sending to 192.168.2.103:5060 (NAT) Using INVITE request as basis request - MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA Found peer ‘3784433’ for ‘3784433’ from 192.168.2.103:5060 \<— Reliably Transmitting (NAT) to 192.168.2.103:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;received=192.168.2.103;rport=5060 From: sip:3784433@192.168.2.101;tag=a76b7847 To: sip:1001213@192.168.2.101;tag=as2ee36d25 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 1 INVITE Server: FPBX-2.9.0(1.8.24.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="03e7d85d" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA’ in 32000 ms (Method: INVITE) \<— SIP read from UDP:192.168.2.103:5060 —\> ACK [sip:1001213@192.168.2.101](mailto:sip:1001213@192.168.2.101) SIP/2.0 Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;rport Max-Forwards: 70 To: sip:1001213@192.168.2.101;tag=as2ee36d25 From: sip:3784433@192.168.2.101;tag=a76b7847 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 1 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — \<— SIP read from UDP:192.168.2.103:5060 —\> INVITE [sip:1001213@192.168.2.101](mailto:sip:1001213@192.168.2.101) SIP/2.0 Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;rport Max-Forwards: 70 Contact: sip:3784433@192.168.2.103:5060 To: sip:1001213@192.168.2.101 From: sip:3784433@192.168.2.101;tag=a76b7847 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5.5 stamp 71236 Authorization: Digest [username=“3784433”,realm=“asterisk”,nonce=“03e7d85d”,uri="sip:1001213@192.168.2.101](mailto:username=%223784433%22,realm=%22asterisk%22,nonce=%2203e7d85d%22,uri=%22sip:1001213@192.168.2.101)",response=“aa2d7c2da92f0a718ebc1153926b3d35”,algorithm=MD5 Content-Length: 306 v=0 o=- 13030640618233466 1 IN IP4 192.168.2.103 s=X-Lite 4 release 4.5.5 stamp 71236 c=IN IP4 192.168.2.103 t=0 0 m=audio 50394 RTP/AVP 125 9 8 0 100 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv \<-------------\> — (14 headers 12 lines) — Sending to 192.168.2.103:5060 (NAT) Using INVITE request as basis request - MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA Found peer ‘3784433’ for ‘3784433’ from 192.168.2.103:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 125 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Found unknown media description format opus for ID 125 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20000100c (ulaw|alaw|speex16|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.2.103:50394 Looking for 1001213 in freeconet1 (domain 192.168.2.101) list\_route: hop: sip:3784433@192.168.2.103:5060 \<— Transmitting (NAT) to 192.168.2.103:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;received=192.168.2.103;rport=5060 From: sip:3784433@192.168.2.101;tag=a76b7847 To: sip:1001213@192.168.2.101 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 2 INVITE Server: FPBX-2.9.0(1.8.24.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1001213@192.168.2.101:5060 Content-Length: 0 \<------------\> – Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000006”, “X-Fid: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA”) in new stack – Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000006”, “CALLERID(num)=48223784433”) in new stack – Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000006”, “SIP/1001213@freeconet-out”) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 – Called SIP/1001213@freeconet-out – SIP/freeconet-out-00000007 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) – Auto fallthrough, channel ‘SIP/3784433-00000006’ status is ‘CONGESTION’ \<— Reliably Transmitting (NAT) to 192.168.2.103:5060 —\> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;received=192.168.2.103;rport=5060 From: sip:3784433@192.168.2.101;tag=a76b7847 To: sip:1001213@192.168.2.101;tag=as1ee85282 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 2 INVITE Server: FPBX-2.9.0(1.8.24.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 \<------------\> – Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000006”, “”) in new stack == Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000006’ \<— SIP read from UDP:192.168.2.103:5060 —\> ACK [sip:1001213@192.168.2.101](mailto:sip:1001213@192.168.2.101) SIP/2.0 Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;rport Max-Forwards: 70 To: sip:1001213@192.168.2.101;tag=as1ee85282 From: sip:3784433@192.168.2.101;tag=a76b7847 Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA CSeq: 2 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Really destroying SIP dialog ‘MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA’ Method: ACK [/quote] ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [December 4, 2013, 2:32pm UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667/4 "2013-12-04T14:32:09Z")
The trace isn’t consistent with the dialplan you provided. It looks like it came from a dialplan with invalid substring syntax, but there is no substringing in the dialplan you provided. My guess is you have ${undefinedvariable}:1:3 rather than ${EXTEN:1:3} ---
**Author:** ![bialyzs3](https://avatars.discourse-cdn.com/v4/letter/b/b19c9b/32.png) [@bialyzs3](https://community.asterisk.org/u/bialyzs3)\ **Post date:** [December 4, 2013, 3:01pm UTC](https://community.asterisk.org/t/asterisk-problems-with-outgoing-calls/44667/5 "2013-12-04T15:01:21Z")
This you mean - exten =\> \_X.,3,Dial(SIP/${undefinedvariable}:1:3@freeconet-out) ? if so, then i show this in terminal: [quote] == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 – Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000000”, “X-Fid: MDIwZDZmNGRiM2Q1NmU5OThiM2Y4OTRiYWQ1ZWYyZjI”) in new stack – Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000000”, “CALLERID(num)=48223784433”) in new stack – Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000000”, “SIP/:1:3@freeconet-out”) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 – Called SIP/:1:3@freeconet-out – SIP/freeconet-out-00000001 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) – Auto fallthrough, channel ‘SIP/3784433-00000000’ status is ‘CONGESTION’ – Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000000”, “”) in new stack == Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000000’ [/quote] ps. I’m just starting their adventure with asterisk 😄