TIPS HiFormance w/sip phone

[ekiMMike]

So I have provisioned a HiFormance server. I have done lots of circular thread reading/following on this site trying to get this to work. I now have scaled back my aspirations of using my obi110 as an ATA and now use an aastra ip phone.

I just verified my configurations against:
http://nerdvittles.com/?p=26632
(Triple Crown article)

An outbound call from my SIP phone results in:
"your call can not be completed as dialed..."

Could someone provide some troubleshooting advice/steps?

 

[ekiMMike]

The crickets are scaring the unicorn...

 

[mjopling]

I am having the same exact issue! I have had success with old versions of FreePBX and Wazo but can not get my new 13-13 to place an outgoing call to any of my different providers. I read about creating a new Time Group and this was also unsuccessful.

Thinking we may have the same problem - hope we hear some from others for what to do next.

My anonymized log script:
Connected to Asterisk 13.22.0 currently running on noreply (pid = 1525)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x40bee10 -- Strict RTP learning after remote address set to: AA.BB.CC.DD:5012
-- Executing [2614XXXXXXX@from-internal:1] Macro("SIP/742-00000008", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/742-00000008", "TOUCH_MONITOR=1534700226.8") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/742-00000008", "AMPUSER=742") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/742-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/742-00000008", "1?Set(REALCALLERIDNUM=742)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/742-00000008", "AMPUSER=742") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/742-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/742-00000008", "AMPUSERCIDNAME=742") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("SIP/742-00000008", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/742-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/742-00000008", "AMPUSERCID=742") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/742-00000008", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/742-00000008", "CALLERID(all)="742" <742>") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/742-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/742-00000008", "1?Set(GROUP(concurrency_limit)=742)") in new stack
-- Executing [s@macro-user-callerid:15] NoOp("SIP/742-00000008", "Macro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/742-00000008", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,17)
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/742-00000008", "1?continue") in new stack
-- Goto (macro-user-callerid,s,35)
-- Executing [s@macro-user-callerid:35] Set("SIP/742-00000008", "CALLERID(number)=742") in new stack
-- Executing [s@macro-user-callerid:36] Set("SIP/742-00000008", "CALLERID(name)=742") in new stack
-- Executing [s@macro-user-callerid:37] GotoIf("SIP/742-00000008", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:38] Set("SIP/742-00000008", "CDR(cnam)=742") in new stack
-- Executing [s@macro-user-callerid:39] Set("SIP/742-00000008", "CDR(cnum)=742") in new stack
-- Executing [s@macro-user-callerid:40] Set("SIP/742-00000008", "CHANNEL(language)=en") in new stack
-- Executing [[2614XXXXXXX@from-internal:2] NoCDR("SIP/742-00000008", "") in new stack
-- Executing [[2614XXXXXXX@from-internal:3] Progress("SIP/742-00000008", "") in new stack
-- Executing [[2614XXXXXXX@from-internal:4] Wait("SIP/742-00000008", "1") in new stack
> 0x40bee10 -- Strict RTP switching to RTP target address AA.BB.CC.DD:5012 as source
-- Executing [[2614XXXXXXX@from-internal:5] Playback("SIP/742-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/742-00000008> Playing 'silence/1.ulaw' (language 'en')
-- <SIP/742-00000008> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- <SIP/742-00000008> Playing 'check-number-dial-again.ulaw' (language 'en')
> 0x40bee10 -- Strict RTP learning complete - Locking on source address AA.BB.CC.DD:5012
-- Executing [[2614XXXXXXX@from-internal:6] Wait("SIP/742-00000008", "1") in new stack
-- Executing [[2614XXXXXXX@from-internal:7] Congestion("SIP/742-00000008", "20") in new stack
[2018-08-19 13:37:14] WARNING[7112][C-00000008]: channel.c:5080 ast_prod: Prodding channel 'SIP/742-00000008' failed
== Spawn extension (from-internal, 2614XXXXXXX, 7) exited non-zero on 'SIP/742-00000008'
-- Executing [h@from-internal:1] Macro("SIP/742-00000008", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/742-00000008", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/742-00000008", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/742-00000008", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/742-00000008' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/742-00000008'

 

[ekiMMike]

Both "misery loves company" and "hope springs eternal" come to mind.

 

[wardmundy]

The crickets are scaring the unicorn...

You're likely to keep hearing crickets unless you post the call progress from the Asterisk CLI. How else could someone respond to "your call cannot be completed as dialed"??

 

[wardmundy]

@mjopling: Is 742 an extension you created? Are you using LEAN or ENCHILADA? Any reason for not trying Enchilada's 701 default extension?? Provide some info about the outbound trunk you are using as well. Is there an outbound route supporting that route and your dial string?

 

[mjopling]

Enchilada on HiFormance

I have multiple extensions, 701 has exactly the same behavior. Vitelity trunks also... I appreciate your looking at this!

Trunk: /General\
Name voipms; Hide caller ID=No; Outbound caller ID set "name" <number>; CID options = allow any; maximum channels = 10; Trunk options=System; continue if busy = no; disable trunk =no

Trunk: /Manipulation rules\
All blank

Trunk: /sip Settings\ /Outgoing\
Name=VoIPmsSFO
Peer details (anonymized):
username=123456_SFO
type=friend
trustrpid=yes
sendrpid=yes
secret=mypassword
qualify=yes
nat=yes
insecure=port,invite
host=chicago3.voip.ms
fromuser=123456_SFO
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw

Trunk: /sip Settings\ /Incoming\
Blank except Register string = 123456_SFO:[email protected]/123456_SFO

Outbound routes:/Route settings\
Name: voipmsSFOroute
CID: <415XXXXXXX>
Override = no
Time Group: TwentyFourHrs
Trunk sequence: VoIPmsSFO
Normal congestion


Outbound routes: /dial patterns\
Prepend, prefix. Match pattern
Blank, 2, 1NXXNXXXXXX \415XXXXXXX
1, 2, NXXNXXXXXX \415XXXXXXX
1614, 2, NXXXXXX \415XXXXXXX

 

[wardmundy]

And what does the Asterisk CLI show when you attempt to make a call??

 

[ekiMMike]

Ward,

The unicorn woke up I guess.
I started the asterisk command line to capture the traffic and called my GV number from an external phone.
The call now goes through.
I made no changes since the last failure.

My guess is Hiformance did something on their end.

We will see if this lasts/remains stable.

 

[mjopling]

My inbound calls work - just the outbounds do not. What you just described was an inbound call. Do your outbounds work? My continue NOT to work.

 

[ekiMMike]

Calls between my cell phone and my GV SIP phone are working in both directions. To add some clarity the GV SIP phone is an extension on the Hiformance server. I do not have a dedicated PBX in addition to the HF box.

My goal is to migrate 2 separate obi110 with devices (and their 2 GV #s - no longer working with GV) to the HF PIAF machine. I will have 2 GV trunks each with a dedicated extension pointed to an obi110 configured as a standard ATA.

 

[ekiMMike]

Getting braver I unplugged my sip phone and configured my obi110 to use the pbx as a sip device.
I am able to receive calls to the obi phone but outbound from the obi phone fail with a 503 error and recording.

Any helpful hints?
the redacted asterisk dialog of the failure is:
------------------
noreply*CLI>
== Setting global variable 'SIPDOMAIN' to 'pbx.domain.org'
-- Executing [18005551212@from-internal:1] ResetCDR("PJSIP/707-0000001d", "") in new stack
-- Executing [18005551212@from-internal:2] NoCDR("PJSIP/707-0000001d", "") in new stack
-- Executing [18005551212@from-internal:3] Progress("PJSIP/707-0000001d", "") in new stack
> 0x7efe7043c5c0 -- Strict RTP learning after remote address set to: 11.2.33..123:16608
-- Executing [18005551212@from-internal:4] Wait("PJSIP/707-0000001d", "1") in new stack
> 0x7efe7043c5c0 -- Strict RTP qualifying stream type: audio
> 0x7efe7043c5c0 -- Strict RTP switching source address to 11.2.33..123:1024
-- Executing [18005551212@from-internal:5] Playback("PJSIP/707-0000001d", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <PJSIP/707-0000001d> Playing 'silence/1.ulaw' (language 'en')
-- <PJSIP/707-0000001d> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- <PJSIP/707-0000001d> Playing 'check-number-dial-again.ulaw' (language 'en')
> 0x7efe7043c5c0 -- Strict RTP learning complete - Locking on source address 11.2.33..123:1024
-- Executing [18005551212@from-internal:6] Wait("PJSIP/707-0000001d", "1") in new stack
-- Executing [18005551212@from-internal:7] Congestion("PJSIP/707-0000001d", "20") in new stack
== Spawn extension (from-internal, 18005551212, 7) exited non-zero on 'PJSIP/707-0000001d'
-- Executing [h@from-internal:1] Macro("PJSIP/707-0000001d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/707-0000001d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/707-0000001d", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/707-0000001d", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/707-0000001d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/707-0000001d'
[2018-08-21 00:52:36] WARNING[5188]: pjproject:0 <?>: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
[2018-08-21 00:52:37] DEBUG[5188]: res_pjsip_outbound_registration.c:625 stateless_send_resolver_callback: Registration using newly created transport 0x7efe5c4e6d88
[2018-08-21 00:52:37] DEBUG[28672]: res_pjsip_outbound_registration.c:1098 save_response_fields_to_client_state: Stored service-route: <sip:redact3WHJDPGNLXMNSOOG6UWCW4GZNSUHOHPSL5FxVK7Y2KJPTFA6OFJB5:5060;uri-econt=redactBI6B3DDARLUx73YNP3ZZYN5ZMQKC3IZ4GAPQJBGE4RKK6PARBRMLDx3CWFYMIJLYE5FZTO6PQLSLLD6WIKARGPSLVWJKWQPR3AVVVV4ZC77PBCY6TFXHKxXGW7XLCB4AK5AVCEQGVAATL6PD57WKxBXBQ;lr>
[2018-08-21 00:52:37] DEBUG[28672]: res_pjsip_outbound_registration.c:1098 save_response_fields_to_client_state: Stored service-route: <sip:redactZXW42N4XTB5RxHWEZX6WAWZN2IBG6IHVRXETHLQEZYKC4TIAWx2X6VU:5060;transport=udp;lr;uri-econt=X5SS6NXOK>
[2018-08-21 00:52:37] DEBUG[28672]: res_pjsip_outbound_registration.c:1104 save_response_fields_to_client_state: Stored associated uri: <sip:redactRGxxDKMBRGQ2DAOBUHEZTANBRGQJBIMBRGA4TMOBQHAYDSMBRGMZDKNJSGU2DI===@obihai.sip.google.com>

==========

 

[mjopling]

Can some one that is familiar with the Asterisk CLI printouts PLEASE give us some hints of how to further troubleshoot this issue?

Does:"-- Executing [[2614XXXXXXX@from-internal:5] Playback("SIP/742-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack" suggest that the Dial Pattern is not stripping the 2 off the Route before transferring to the Trunk ?

 

[mjopling]

is there additional information I could provide that could help answer this issue?

 

[wardmundy]

Can some one that is familiar with the Asterisk CLI printouts PLEASE give us some hints of how to further troubleshoot this issue?

Does:"-- Executing [[2614XXXXXXX@from-internal:5] Playback("SIP/742-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack" suggest that the Dial Pattern is not stripping the 2 off the Route before transferring to the Trunk ?

Correct. The 2 should be in the Prefix field if you want Asterisk to strip it off before sending the call to the trunk.

 

[mjopling]

Ward, I know how the prefix works but trying to reconcile what I see in the CLI log to determine where the problem arises that interrupts the call from actually going out a trunk. The third post in this thread contains a more complete anonymized CLI log script of which the most pertinent part is possibly:
-- Executing [s@macro-user-callerid:40] Set("SIP/742-00000008", "CHANNEL(language)=en") in new stack
-- Executing [[2614XXXXXXX@from-internal:2] NoCDR("SIP/742-00000008", "") in new stack
-- Executing [[2614XXXXXXX@from-internal:3] Progress("SIP/742-00000008", "") in new stack
-- Executing [[2614XXXXXXX@from-internal:4] Wait("SIP/742-00000008", "1") in new stack
> 0x40bee10 -- Strict RTP switching to RTP target address AA.BB.CC.DD:5012 as source
-- Executing [[2614XXXXXXX@from-internal:5] Playback("SIP/742-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack

I have reinstalled, tried multiple different outbound routes, call patterns (with and without 1 prefix), extensions, trunks, vendors.... All inbound calls and internal calls work fine. GV, Vitelity, one VOIP.ms outbounds have always failed

 

[wardmundy]

Keep in mind that outbound routes get processed from the top to the bottom. Try moving the correct route UP in the hierarchy.

 

[ekiMMike]

I posted my asterisk log back on the 20th as requested.
Has anyone any guidance towards resolution or other diagnostics to perform?