No Audio and getting the notice "translate.c: x lost frame(s)"

scolastico · March 8, 2021, 5:52am

Hello i have no adio in booth directions (internal and external). I use following configuration for chan_sip:

username=my number
type=peer
secret=my password
qualify=yes
nat=yes
insecure=invite
host=192.168.178.1
fromuser=my number
fromdomain=fritz.box
dtmfmode=rfc2833
disallow=all
directmedia=no
context=from-trunk
allow=g722&g726&g729

( i tryd it with multiple codecs but nothing is working )

i think the problem could be this notice: “translate.c: 123 lost frame(s) 124/0 (slin@8000)->(ulaw@8000)”

complete log files: (i censored the numbers out)

5139	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'udptl' (UDPTL)	
5140	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjproject.so' (PJPROJECT Log and Utility Support)	
5141	[2021-03-08 05:36:12] ERROR[13738] res_sorcery_config.c: Unable to load config file 'pjproject.conf'	
5142	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip.so' (Basic SIP resource)	
5143	[2021-03-08 05:36:12] ERROR[11871] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5144	[2021-03-08 05:36:12] ERROR[11871] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5145	[2021-03-08 05:36:12] ERROR[11871] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5146	[2021-03-08 05:36:12] ERROR[11871] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5147	[2021-03-08 05:36:12] NOTICE[11871] sorcery.c: Type 'system' is not reloadable, maintaining previous values	
5148	[2021-03-08 05:36:12] ERROR[11871] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5149	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_authenticator_digest.so' (PJSIP authentication resource)	
5150	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_endpoint_identifier_ip.so' (PJSIP IP endpoint identifier)	
5151	[2021-03-08 05:36:12] ERROR[13738] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5152	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_resolver_unbound.so' (Unbound DNS Resolver Support)	
5153	[2021-03-08 05:36:12] ERROR[13738] config_options.c: Unable to load config file 'resolver_unbound.conf'	
5154	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_stir_shaken.so' (STIR/SHAKEN Module for Asterisk)	
5155	[2021-03-08 05:36:12] ERROR[13738] res_sorcery_config.c: Unable to load config file 'stir_shaken.conf'	
5156	[2021-03-08 05:36:12] ERROR[13738] res_sorcery_config.c: Unable to load config file 'stir_shaken.conf'	
5157	[2021-03-08 05:36:12] ERROR[13738] res_sorcery_config.c: Unable to load config file 'stir_shaken.conf'	
5158	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_musiconhold.so' (Music On Hold Resource)	
5159	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_smdi.so' (Simplified Message Desk Interface (SMDI) Resource)	
5160	[2021-03-08 05:36:12] NOTICE[13738] res_smdi.c: Unable to reload config smdi.conf: SMDI untouched	
5161	[2021-03-08 05:36:12] WARNING[13738] res_smdi.c: No SMDI interfaces were specified to listen on, not starting SDMI listener.	
5162	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)	
5163	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_outbound_publish.so' (PJSIP Outbound Publish Support)	
5164	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_rtp_asterisk.so' (Asterisk RTP Stack)	
5165	[2021-03-08 05:36:12] VERBOSE[13738] res_rtp_asterisk.c: RTP Allocating from port range 10000 -> 20000	
5166	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_publish_asterisk.so' (PJSIP Asterisk Event PUBLISH Support)	
5167	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_mwi.so' (PJSIP MWI resource)	
5168	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))	
5169	[2021-03-08 05:36:12] VERBOSE[1377] chan_sip.c: Reloading SIP	
5170	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))	
5171	[2021-03-08 05:36:12] VERBOSE[1377] netsock2.c: Using SIP TOS bits 96	
5172	[2021-03-08 05:36:12] VERBOSE[1377] netsock2.c: Using SIP CoS mark 4	
5173	[2021-03-08 05:36:12] NOTICE[13738] iax2/provision.c: No IAX provisioning configuration found, IAX provisioning disabled.	
5174	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_fax.so' (Generic FAX Applications)	
5175	[2021-03-08 05:36:12] NOTICE[13738] res_fax.c: Configuration file 'res_fax.conf' not found, not changing options.	
5176	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_adsi.so' (ADSI Resource)	
5177	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_ari.so' (Asterisk RESTful Interface)	
5178	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Registration Support)	
5179	[2021-03-08 05:36:12] ERROR[13738] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'	
5180	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)	
5181	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'codec_speex.so' (Speex Coder/Decoder)	
5182	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'app_confbridge.so' (Conference Bridge Application)	
5183	[2021-03-08 05:36:12] NOTICE[13738] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge	
5184	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'cel_odbc.so' (ODBC CEL backend)	
5185	[2021-03-08 05:36:12] WARNING[13738] cel_odbc.c: Unable to load cel_odbc.conf. No ODBC CEL records!	
5186	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))	
5187	[2021-03-08 05:36:12] WARNING[13738] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages	
5188	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'res_http_post.so' (HTTP POST support)	
5189	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'app_playback.so' (Sound File Playback Application)	
5190	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'codec_opus_open_source.so' (Opus Coder/Decoder)	
5191	[2021-03-08 05:36:12] VERBOSE[13738] loader.c: Reloading module 'app_queue.so' (True Call Queueing)	
5192	[2021-03-08 05:36:12] NOTICE[13738] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules	
5193	[2021-03-08 05:36:12] VERBOSE[13738] asterisk.c: Remote UNIX connection disconnected	
5194	[2021-03-08 05:36:27] VERBOSE[1377][C-0000000f] netsock2.c: Using SIP RTP TOS bits 184	
5195	[2021-03-08 05:36:27] VERBOSE[1377][C-0000000f] netsock2.c: Using SIP RTP CoS mark 5	
5196	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [**my number**@from-trunk:1] NoOp("SIP/**my number**-00000008", "Catch-All DID Match - Found **my number** - You probably want a DID for this.") in new stack	
5197	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [**my number**@from-trunk:2] Set("SIP/**my number**-00000008", "__FROM_DID=**my number**") in new stack	
5198	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [**my number**@from-trunk:3] Goto("SIP/**my number**-00000008", "ext-did,s,1") in new stack	
5199	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (ext-did,s,1)	
5200	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:1] Set("SIP/**my number**-00000008", "__DIRECTION=INBOUND") in new stack	
5201	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:2] Gosub("SIP/**my number**-00000008", "sub-record-check,s,1(in,s,dontcare)") in new stack	
5202	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:1] GotoIf("SIP/**my number**-00000008", "0?initialized") in new stack	
5203	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:2] Set("SIP/**my number**-00000008", "__REC_STATUS=INITIALIZED") in new stack	
5204	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:3] Set("SIP/**my number**-00000008", "NOW=1615178187") in new stack	
5205	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:4] Set("SIP/**my number**-00000008", "__DAY=08") in new stack	
5206	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:5] Set("SIP/**my number**-00000008", "__MONTH=03") in new stack	
5207	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:6] Set("SIP/**my number**-00000008", "__YEAR=2021") in new stack	
5208	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:7] Set("SIP/**my number**-00000008", "__TIMESTR=20210308-053627") in new stack	
5209	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:8] Set("SIP/**my number**-00000008", "__FROMEXTEN=unknown") in new stack	
5210	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:9] Set("SIP/**my number**-00000008", "__MON_FMT=wav") in new stack	
5211	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:10] NoOp("SIP/**my number**-00000008", "Recordings initialized") in new stack	
5212	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:11] ExecIf("SIP/**my number**-00000008", "0?Set(ARG3=dontcare)") in new stack	
5213	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:12] Set("SIP/**my number**-00000008", "REC_POLICY_MODE_SAVE=") in new stack	
5214	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:13] ExecIf("SIP/**my number**-00000008", "0?Set(REC_STATUS=NO)") in new stack	
5215	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:14] GotoIf("SIP/**my number**-00000008", "2?checkaction") in new stack	
5216	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (sub-record-check,s,17)	
5217	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@sub-record-check:17] GotoIf("SIP/**my number**-00000008", "1?sub-record-check,in,1") in new stack	
5218	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (sub-record-check,in,1)	
5219	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [in@sub-record-check:1] NoOp("SIP/**my number**-00000008", "Inbound Recording Check to s") in new stack	
5220	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [in@sub-record-check:2] Set("SIP/**my number**-00000008", "FROMEXTEN=unknown") in new stack	
5221	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [in@sub-record-check:3] ExecIf("SIP/**my number**-00000008", "12?Set(FROMEXTEN=**my number**)") in new stack	
5222	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [in@sub-record-check:4] Gosub("SIP/**my number**-00000008", "recordcheck,1(dontcare,in,s)") in new stack	
5223	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("SIP/**my number**-00000008", "Starting recording check against dontcare") in new stack	
5224	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("SIP/**my number**-00000008", "dontcare") in new stack	
5225	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)	
5226	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [recordcheck@sub-record-check:3] Return("SIP/**my number**-00000008", "") in new stack	
5227	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [in@sub-record-check:5] Return("SIP/**my number**-00000008", "") in new stack	
5228	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:3] Set("SIP/**my number**-00000008", "CHANNEL(tonezone)=us") in new stack	
5229	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:4] ExecIf("SIP/**my number**-00000008", "0?Set(__FROM_DID=s)") in new stack	
5230	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:5] Set("SIP/**my number**-00000008", "CDR(did)=**my number**") in new stack	
5231	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:6] ExecIf("SIP/**my number**-00000008", "0 ?Set(CALLERID(name)=**my number**)") in new stack	
5232	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:7] Set("SIP/**my number**-00000008", "__MOHCLASS=") in new stack	
5233	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:8] Set("SIP/**my number**-00000008", "__REVERSAL_REJECT=FALSE") in new stack	
5234	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:9] GotoIf("SIP/**my number**-00000008", "1?post-reverse-charge") in new stack	
5235	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (ext-did,s,11)	
5236	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:11] NoOp("SIP/**my number**-00000008", "") in new stack	
5237	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:12] Set("SIP/**my number**-00000008", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack	
5238	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:13] Set("SIP/**my number**-00000008", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack	
5239	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:14] Set("SIP/**my number**-00000008", "CALLERID(name-pres)=allowed_not_screened") in new stack	
5240	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:15] Set("SIP/**my number**-00000008", "CALLERID(num-pres)=allowed_not_screened") in new stack	
5241	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:16] NoOp("SIP/**my number**-00000008", "CallerID Entry Point") in new stack	
5242	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ext-did:17] Goto("SIP/**my number**-00000008", "app-announcement-3,s,1") in new stack	
5243	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (app-announcement-3,s,1)	
5244	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:1] GotoIf("SIP/**my number**-00000008", "0?begin") in new stack	
5245	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:2] Answer("SIP/**my number**-00000008", "") in new stack	
5246	[2021-03-08 05:36:27] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:3] Wait("SIP/**my number**-00000008", "1") in new stack	
5247	[2021-03-08 05:36:28] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:4] NoOp("SIP/**my number**-00000008", "Playing announcement record-question") in new stack	
5248	[2021-03-08 05:36:28] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:5] Playback("SIP/**my number**-00000008", "custom/record-question,noanswer") in new stack	
5249	[2021-03-08 05:36:28] VERBOSE[13839][C-0000000f] file.c: <SIP/**my number**-00000008> Playing 'custom/record-question.slin' (language 'de_DE')	
5250	[2021-03-08 05:36:29] NOTICE[13839][C-0000000f] translate.c: 123 lost frame(s) 124/0 (slin@8000)->(ulaw@8000)	
5251	[2021-03-08 05:36:33] NOTICE[13839][C-0000000f] translate.c: 316 lost frame(s) 317/0 (slin@8000)->(ulaw@8000)	
5252	[2021-03-08 05:36:39] NOTICE[13839][C-0000000f] translate.c: 620 lost frame(s) 621/0 (slin@8000)->(ulaw@8000)	
5253	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@app-announcement-3:6] Goto("SIP/**my number**-00000008", "ivr-1,s,1") in new stack	
5254	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (ivr-1,s,1)	
5255	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:1] Set("SIP/**my number**-00000008", "_IVR_CONTEXT_ivr-1=") in new stack	
5256	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:2] Set("SIP/**my number**-00000008", "_IVR_CONTEXT=ivr-1") in new stack	
5257	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:3] Set("SIP/**my number**-00000008", "__IVR_RETVM=") in new stack	
5258	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:4] GotoIf("SIP/**my number**-00000008", "1?skip") in new stack	
5259	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx_builtins.c: Goto (ivr-1,s,6)	
5260	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:6] Set("SIP/**my number**-00000008", "IVR_MSG=") in new stack	
5261	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:7] Set("SIP/**my number**-00000008", "TIMEOUT(digit)=3") in new stack	
5262	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] func_timeout.c: Digit timeout set to 3.000	
5263	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:8] ExecIf("SIP/**my number**-00000008", "0?Background()") in new stack	
5264	[2021-03-08 05:36:41] VERBOSE[13839][C-0000000f] pbx.c: Executing [s@ivr-1:9] WaitExten("SIP/**my number**-00000008", "15,") in new stack	
5265	[2021-03-08 05:36:42] VERBOSE[13839][C-0000000f] pbx.c: Spawn extension (ivr-1, s, 9) exited non-zero on 'SIP/**my number**-00000008'	
5266	[2021-03-08 05:36:42] VERBOSE[13839][C-0000000f] pbx.c: Executing [h@ivr-1:1] Hangup("SIP/**my number**-00000008", "") in new stack	
5267	[2021-03-08 05:36:42] VERBOSE[13839][C-0000000f] pbx.c: Spawn extension (ivr-1, h, 1) exited non-zero on 'SIP/**my number**-00000008'

dicko · March 8, 2021, 5:57am

Replace

allow=g722&g726&g729

With

allow=all

And see if a codec can be negotiated

scolastico · March 8, 2021, 6:29am

looks like the notice is gone but still no audio and the call hangs up around 45sec.

Stewart1 · March 8, 2021, 6:41am

In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config you must also restart Asterisk.

If you still have trouble, at the Asterisk command prompt type
sip set debug on
If your extensions are pjsip, also type
pjsip set logger on
make a test call, paste the Asterisk log for the call (which will now include SIP traces) at pastebin.freepbx.org and post the link here.

scolastico · March 9, 2021, 4:08am

Done: http://pastebin.freepbx.org/view/ab259c53
(New users cant write urls down here so replace the dot with a real “.”.)

[you should be able to post links now - mod]

Stewart1 · March 9, 2021, 4:54am

How is FreePBX connected to the Fritzbox? If it is directly connected to a LAN port and gets an address like 192.168.178.20 (and I assume that what you replaced with **free pbx ip** is a public IP), then it appears that Local Networks is not correctly set. Please confirm that in Asterisk SIP Settings, you have Local Networks set to 192.168.178.0 / 24 (or whatever your LAN needs) and that you have restarted Asterisk after saving this setting.

If you have a more complex network, please describe all elements (routers, firewalls, etc.) between the PBX, the Fritzbox and the internet. If your PBX has more than one NIC, provide details.

scolastico · March 9, 2021, 7:31pm

its a local network with a raspberry pi and a normal local ip (192.168.178.XXX) the network is normaly setup throug the fritzbox and has no abnormal modifactions. only the rasp has a static local ip

scolastico · March 10, 2021, 3:49am

I tested it now out with a server installation and a static IPv4 so it can run without the local network and connected it with sip to my service provider directly without the fritz.box but now i cant recive calls and if i make calls they get disconnected after 45 sec. with the message:

Disconnecting channel 'PJSIP/50-00000002' for lack of audio RTP activity in 35 seconds

Log files: https://haste.scolasti.co/ilaneledur.md

Stewart1 · March 10, 2021, 4:53am

Unfortunately, without a SIP trace, the log you posted has few clues.

First, in Asterisk SIP Settings, confirm that both External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config you must restart Asterisk.

If you still have trouble, find the simplest thing that fails. Can you call *43 (echo test) and hear both the instructions and your voice? If not, we’ll debug that first.

If the echo test works, try a call from one extension to another. If it fails, debug that.

If both of the above are ok, we’ll use an outbound call for testing.

At the Asterisk command prompt, type
pjsip set logger on
and make a test call. Paste the log (which will now include a SIP trace) and post the link.

scolastico · March 10, 2021, 8:50pm

yeah the problem seems to be an invalid setup firewall and some network issues after some restarts and try’n error it worked how it should be on the server installation (the raspbx seems to have also network issues but i cant publish it with a static ip because my provider gives static ips only for buisniss networks and not normal home network) idk completly how i got it to work but it works

Thank you @Stewart1 and @dicko for the great help.

system · June 3, 2021, 11:07pm

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