# You have reached a non-working number **URL:** \ **Category:** Asterisk Support\ **Created:** [March 23, 2012, 5:32am UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667 "2012-03-23T05:32:18Z")\ **Posts on this page:** 8\ **Page:** 1
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [March 23, 2012, 5:32am UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/1 "2012-03-23T05:32:18Z")
Hello, I’m running Asterisk 1.8.4.4 I’ve had it up and running for several weeks now. But, for some reason, when I call in, using my DID number, I get the message “you have reached a non-working number, 14 switch 78-4”. My DID number seems to be working correctly. I rebooted my system, which didn’t help. I’m a quite lost as to what’s causing the problem, to try and even problem solve. Any suggestions or thoughts? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [March 23, 2012, 7:22am UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/2 "2012-03-23T07:22:22Z")
That message isn’t coming from Asterisk. What do you mean by your “DID” is working correctly? You seem to have said it is not in the previous paragraph! You need to start by providing a verbose CLI trace, showing how the call is being handled and may need to provide a technology specific protocol trace. I’m guessing that you are using SIP, in which case you need to use “sip set debug on”. Although probably not relevant here, by DID do you mean a PSTN trunk on which dialed digits are forwarded (the traditional definition), or do you just mean an incoming line for calls that originated on the PSTN (the loose usage in the SIP community)? ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [March 23, 2012, 2:08pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/3 "2012-03-23T14:08:25Z")
The calls originate on PSTN. I bought a telephone number that people call to reach my Asterisk system. I do not have any specific phone models associated with the system. It’s just one number. I turned on sip debug. This is what I received --\> [Mar 23 08:48:40] NOTICE[1449]: chan\_sip.c:12403 sip\_reregister: – Re-registration for [3476479911@sip.broadvoice.com](mailto:3476479911@sip.broadvoice.com) REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 206.15.156.221:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK207a1734;rport Max-Forwards: 70 From: sip:3476479911@sip.broadvoice.com;tag=as112889dc To: sip:3476479911@sip.broadvoice.com Call-ID: [47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com](mailto:47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com) CSeq: 1415 REGISTER User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Authorization: Digest username=“3476479911”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXh04tfoxgTpt2moxBW”, response=“a1e7c3c8e82903ff3553adc233621bf1”, qop=auth, cnonce=“17edf463”, nc=0000049d Expires: 120 Contact: sip:1001@108.166.89.184:5060 Content-Length: 0 * * * \<— SIP read from UDP:206.15.156.221:5060 —\> SIP/2.0 200 OK Call-ID: [47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com](mailto:47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com) CSeq: 1415 REGISTER From: sip:3476479911@sip.broadvoice.com;tag=as112889dc To: sip:3476479911@sip.broadvoice.com Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK207a1734;rport=5060 Contact: sip:1001@108.166.89.184:5060 Expires: 30 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Scheduling destruction of SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ in 6400 ms (Method: REGISTER) [Mar 23 08:48:40] NOTICE[1449]: chan\_sip.c:19793 handle\_response\_register: Outbound Registration: Expiry for [sip.broadvoice.com](http://sip.broadvoice.com) is 30 sec (Scheduling reregistration in 24 s) Really destroying SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ Method: REGISTER [Mar 23 08:49:04] NOTICE[1449]: chan\_sip.c:12403 sip\_reregister: – Re-registration for [3476479911@sip.broadvoice.com](mailto:3476479911@sip.broadvoice.com) REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 206.15.156.221:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1cdb7da6;rport Max-Forwards: 70 From: sip:3476479911@sip.broadvoice.com;tag=as447e1396 To: sip:3476479911@sip.broadvoice.com Call-ID: [47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com](mailto:47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com) CSeq: 1416 REGISTER User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Authorization: Digest username=“3476479911”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXh04tfoxgTpt2moxBW”, response=“98da3de4f1ad179f3121cec8a4d15f18”, qop=auth, cnonce=“24680e00”, nc=0000049e Expires: 120 Contact: sip:1001@108.166.89.184:5060 Content-Length: 0 * * * \<— SIP read from UDP:206.15.156.221:5060 —\> SIP/2.0 200 OK Call-ID: [47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com](mailto:47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com) CSeq: 1416 REGISTER From: sip:3476479911@sip.broadvoice.com;tag=as447e1396 To: sip:3476479911@sip.broadvoice.com Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1cdb7da6;rport=5060 Contact: sip:1001@108.166.89.184:5060 Expires: 30 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Scheduling destruction of SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ in 6400 ms (Method: REGISTER) [Mar 23 08:49:04] NOTICE[1449]: chan\_sip.c:19793 handle\_response\_register: Outbound Registration: Expiry for [sip.broadvoice.com](http://sip.broadvoice.com) is 30 sec (Scheduling reregistration in 24 s) Really destroying SIP dialog '47dc840c21ef00b04e6cae2a4e6150b1@sip.broadvoice.com’ Method: REGISTER This is what I have in my sip —\> [general] context = unauthenticated ; default context for incoming calls allowguest = no ; disable unauthenticated calls srvlookup = yes ; enable DNS SRV record lookup on outbound calls udpbindaddr = 0.0.0.0 ; listen for UDP requests on all interfaces tcpenable = no ; disable TCP support pedantic = no ;overrideswitch = yes register =\> [3476479911@sip.broadvoice.com](mailto:3476479911@sip.broadvoice.com):passwd:3476479911@sip.broadvoice.com/1001 ; [broadvoice.com](http://broadvoice.com) [office-phone] (!) ; create a template for our devices context = incoming type = friend ; the channel driver will match on username first, IP second host = dynamic ; the device will register with asterisk nat = yes ; assume the device is behind a NAT qualify = yes canreinvite = no secret = passwd dtmfmode = rfc2833 disallow = all ; reset which voice codes this device will accept or offer allow = ulaw ; which audio codecs to accept from, and request to, the device allow = alaw 1001 ; seems that this is not only related to soft phone. needed an ext for DID. didn’t lik$ type=friend username=1001 secret=passwd host=dynamic nat=yes qualify=yes dtmfmode = rfc2833 ; changed dtmfmode, disallow, allow, dtfm and mapped an ext to DID #, went through despit$ disallow = all allow = ulaw dtfm = inband sip.broadvoice.com type = peer user = phone host = [sip.broadvoice.com](http://sip.broadvoice.com) context = incoming fromdomain = [sip.broadvoice.com](http://sip.broadvoice.com) fromuser = 3476479911 secret = passwd2 username = 3476479911 insecure = invite authname = 3476479911 dtmfmode = rfc2833 ; changed dtmfmode, disallow, allow, dtfm and mapped an ext to DID #, went through despite wha$ disallow = all allow = ulaw dtfm = inband canreinvite = no ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [March 23, 2012, 2:36pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/4 "2012-03-23T14:36:12Z")
Does this mean that INVITEs never reach you, or did you just not include an example? In the former case, is 108.166.89.184 your correct public IP address and is there unconditional port forwarding for 5060/UDP? In the latter case, you will need to provide the relevant trace and versbose console output. Is there an extension 1001 in the incoming context? You don’t appear to have direct in dialing, but that is probably not relevant. As noted, the Asterisk community misuses “DID”. ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [March 23, 2012, 5:04pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/5 "2012-03-23T17:04:46Z")
Hello, I put “bindport = 5060” in my sip configuration, and was able to get back through. If my IP address for my Asterisk server is unchanging, should I also make that the bindaddr? Or, should that be the IP address of where the calls are getting forwarded from? By INVITEs, are you referring to the “canreinvite’s”? I’ve been reading over the voipinfo wiki… ([voip-info.org/wiki/view/Aste … anreinvite](http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite)). I’m behind a nat, so I figured I should turn that off. Or should I use “nonat”? Or maybe I’m missing your question about INVITEs all together… I’m new to Asterisk. And, I appreciate your help and directives!! ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [March 23, 2012, 5:09pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/6 "2012-03-23T17:09:12Z")
bindaddr, if used, should always be the address of a local interface or 0.0.0.0. canreinvite is an obsolete name for directmedia; I was talking about the INVITE that starts an incoming call, not a re-INVITE. If a call reaches your machine, the first you will see of it is an incoming SIP INVITE method packet. ---
**Author:** ![KBennett](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kbennett/32/146_2.png) [@KBennett](https://community.asterisk.org/u/KBennett)\ **Post date:** [March 23, 2012, 5:22pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/7 "2012-03-23T17:22:36Z")
This is what I have in the beginning of a call from the command line: == Using SIP RTP CoS mark 5 SIP Debugging enabled [Mar 23 12:12:07] NOTICE[1449]: chan\_sip.c:12403 sip\_reregister: – Re-registration for [3476479911@sip.broadvoice.com](mailto:3476479911@sip.broadvoice.com) REGISTER 10 headers, 0 lines Reliably Transmitting (NAT) to 206.15.156.221:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK4c4f0b99;rport Max-Forwards: 70 From: sip:3476479911@sip.broadvoice.com;tag=as44c55872 To: sip:3476479911@sip.broadvoice.com Call-ID: [26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com](mailto:26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com) CSeq: 157 REGISTER User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Expires: 120 Contact: sip:1001@108.166.89.184:5060 Content-Length: 0 * * * \<— SIP read from UDP:206.15.156.221:5060 —\> SIP/2.0 200 OK Call-ID: [26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com](mailto:26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com) CSeq: 157 REGISTER From: sip:3476479911@sip.broadvoice.com;tag=as44c55872 To: sip:3476479911@sip.broadvoice.com Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK4c4f0b99;rport=5060 Contact: sip:1001@108.166.89.184:5060 Expires: 30 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Scheduling destruction of SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ in 6400 ms (Method: REGISTER) [Mar 23 12:12:07] NOTICE[1449]: chan\_sip.c:19793 handle\_response\_register: Outbound Registration: Expiry for [sip.broadvoice.com](http://sip.broadvoice.com) is 30 sec (Scheduling reregistration in 24 s) \<— SIP read from UDP:206.15.156.221:5060 —\> INVITE sip:3476479911@108.166.89.184:5060 SIP/2.0 Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 1 INVITE From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184 Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857- Contact: sip:3122083026@206.15.156.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 310 Content-Type: application/sdp v=0 o=3457129711 10 10 IN IP4 206.15.156.239 s=- c=IN IP4 206.15.156.239 t=0 0 m=audio 26582 RTP/AVP 0 8 18 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 \<-------------\> — (11 headers 14 lines) — Sending to 206.15.156.221:5060 (no NAT) Using INVITE request as basis request - [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) Found peer ‘[sip.broadvoice.com](http://sip.broadvoice.com)’ for ‘3122083026’ from 206.15.156.221:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format iLBC for ID 96 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x150c (ulaw|alaw|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 206.15.156.239:26582 Looking for 3476479911 in incoming (domain 108.166.89.184:5060) list\_route: hop: sip:3122083026@206.15.156.221:5060 \<— Transmitting (NAT) to 206.15.156.221:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060 From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184 Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:3476479911@108.166.89.184:5060 Content-Length: 0 \<------------\> – Executing [3476479911@incoming:1] Goto(“SIP/sip.broadvoice.com-00000003”, “main\_menu,start,1”) in new stack – Goto (main\_menu,start,1) – Executing [start@main\_menu:1] Set(“SIP/sip.broadvoice.com-00000003”, “step1count=0”) in new stack – Executing [start@main\_menu:2] Answer(“SIP/sip.broadvoice.com-00000003”, “”) in new stack Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (NAT) to 206.15.156.221:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060 From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:3476479911@108.166.89.184:5060 Content-Type: application/sdp Content-Length: 252 v=0 o=root 799446011 799446011 IN IP4 108.166.89.184 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 108.166.89.184 t=0 0 m=audio 18886 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv \<------------\> Retransmitting #1 (NAT) to 206.15.156.221:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 206.15.156.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.156.221V5060-0-541433049-726159298-1332522727857-;received=206.15.156.221;rport=5060 From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:3476479911@108.166.89.184:5060 Content-Type: application/sdp Content-Length: 252 v=0 o=root 799446011 799446011 IN IP4 108.166.89.184 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 108.166.89.184 t=0 0 m=audio 18886 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:206.15.156.221:5060 —\> ACK sip:3476479911@108.166.89.184:5060 SIP/2.0 Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 1 ACK From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde Via: SIP/2.0/UDP 206.15.156.221:5060 Max-Forwards: 69 Content-Length: 0 \<-------------\> — (8 headers 0 lines) — – Executing [start@main\_menu:3] BackGround(“SIP/sip.broadvoice.com-00000003”, “/usr/share/asterisk/sounds/en/dp\_greeting”) in new stack – \ Playing ‘/usr/share/asterisk/sounds/en/dp\_greeting.gsm’ (language ‘en’) Really destroying SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ Method: REGISTER – Executing [start@main\_menu:4] BackGround(“SIP/sip.broadvoice.com-00000003”, “/usr/share/asterisk/sounds/en/dp\_mainMenu”) in new stack – \ Playing ‘/usr/share/asterisk/sounds/en/dp\_mainMenu.gsm’ (language ‘en’) – Executing [start@main\_menu:5] WaitExten(“SIP/sip.broadvoice.com-00000003”, “8”) in new stack \<— SIP read from UDP:206.15.156.221:5060 —\> BYE sip:3476479911@108.166.89.184:5060 SIP/2.0 Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 2 BYE From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde Via: SIP/2.0/UDP 206.15.156.221:5060 Max-Forwards: 69 Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Sending to 206.15.156.221:5060 (NAT) Scheduling destruction of SIP dialog ‘770197-77@206.15.156.221’ in 6400 ms (Method: BYE) \<— Transmitting (NAT) to 206.15.156.221:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 206.15.156.221:5060;received=206.15.156.221;rport=5060 From: sip:3122083026@206.15.156.221;user=phone;tag=cefg To: "katherine bennett"sip:1001@108.166.89.184;tag=as17983dde Call-ID: [770197-77@206.15.156.221](mailto:770197-77@206.15.156.221) CSeq: 2 BYE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 \<------------\> == Spawn extension (main\_menu, start, 5) exited non-zero on ‘SIP/sip.broadvoice.com-00000003’ – Executing [h@main\_menu:1] Set(“SIP/sip.broadvoice.com-00000003”, “result=”) in new stack – Executing [h@main\_menu:2] NoOp(“SIP/sip.broadvoice.com-00000003”, "result is ") in new stack – Executing [h@main\_menu:3] Hangup(“SIP/sip.broadvoice.com-00000003”, “”) in new stack == Spawn extension (main\_menu, h, 3) exited non-zero on 'SIP/sip.broadvoice.com-00000003’ Reliably Transmitting (NAT) to 206.15.156.221:5060: OPTIONS sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK51771e28;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@108.166.89.184;tag=as63decaeb To: sip:sip.broadvoice.com Contact: sip:asterisk@108.166.89.184:5060 Call-ID: 6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Date: Fri, 23 Mar 2012 17:12:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:206.15.156.221:5060 —\> SIP/2.0 200 OK Call-ID: 6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060 CSeq: 102 OPTIONS From: “asterisk” sip:asterisk@108.166.89.184;tag=as63decaeb To: sip:sip.broadvoice.com Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK51771e28;rport=5060 Supported: 100rel Max-Forwards: 70 Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Content-Length: 0 \<-------------\> — (14 headers 0 lines) — Really destroying SIP dialog ‘6e2a58966a1c3be145291e3e3b48ce05@108.166.89.184:5060’ Method: OPTIONS [Mar 23 12:12:31] NOTICE[1449]: chan\_sip.c:12403 sip\_reregister: – Re-registration for [3476479911@sip.broadvoice.com](mailto:3476479911@sip.broadvoice.com) REGISTER 10 headers, 0 lines Reliably Transmitting (NAT) to 206.15.156.221:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1918c69d;rport Max-Forwards: 70 From: sip:3476479911@sip.broadvoice.com;tag=as7f5217b8 To: sip:3476479911@sip.broadvoice.com Call-ID: [26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com](mailto:26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com) CSeq: 158 REGISTER User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Expires: 120 Contact: sip:1001@108.166.89.184:5060 Content-Length: 0 * * * \<— SIP read from UDP:206.15.156.221:5060 —\> SIP/2.0 200 OK Call-ID: [26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com](mailto:26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com) CSeq: 158 REGISTER From: sip:3476479911@sip.broadvoice.com;tag=as7f5217b8 To: sip:3476479911@sip.broadvoice.com Via: SIP/2.0/UDP 108.166.89.184:5060;branch=z9hG4bK1918c69d;rport=5060 Contact: sip:1001@108.166.89.184:5060 Expires: 30 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Scheduling destruction of SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ in 6400 ms (Method: REGISTER) [Mar 23 12:12:31] NOTICE[1449]: chan\_sip.c:19793 handle\_response\_register: Outbound Registration: Expiry for [sip.broadvoice.com](http://sip.broadvoice.com) is 30 sec (Scheduling reregistration in 24 s) Really destroying SIP dialog ‘770197-77@206.15.156.221’ Method: BYE Really destroying SIP dialog '26ee92fb180a30f40c7723d63d6840bc@sip.broadvoice.com’ Method: REGISTER Is this what you were referring to? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [March 23, 2012, 5:32pm UTC](https://community.asterisk.org/t/you-have-reached-a-non-working-number/38667/8 "2012-03-23T17:32:52Z")
That’s an INVITE, but it clearly shows a call to a working number, albeit one on a switch with an excessive packet loss rate. It is possible to send a BYE to abort a call with unacceptable codecs on the OK, but there seems to have been a sufficient delay between the ACK and the BYE that I don’t think that is the case. Maybe the firewall is blocking RTP and it is objecting to that. You will have to ask them what can cause the diagnostic your caller received.