Sip not wotking in Asterisk server

Dear all,
Actually Iam very new to Asterisk, By depending on books, internet and also asterisk community support I got some knowledge on asterisk and succeed to through the FXS and FXO calls to work, using DIGIUM CARDS. Thanks allot to all for your support on this regard.

Now Iam working on SIP based IP Phones (Grandstream) to register in Asterisk, Tried allot but the following issues are getting in this process. please suggest.

root@localhost ~]# asterisk -vvvr
Asterisk 17.4.0, Copyright © 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 17.4.0 currently running on localhost (pid = 3968)
localhost*CLI>

localhostCLI> sip show peers
No such command ‘sip show peers’ (type ‘core show help sip show’ for other possible commands)
localhost
CLI>

localhostCLI> core show help sip show
No such command ‘sip show’.
localhost
CLI>

localhostCLI> module show like sip
Module Description Use Count Status Support Level
app_adsiprog.so Asterisk ADSI Programming Application 0 Running deprecated
1 modules loaded
localhost
CLI>

localhostCLI> pjsip show endpoints
No such command ’ pjsip show endpoints’ (type ‘core show help pjsip show endpoints’ for other possible commands)
localhost
CLI>

localhostCLI> core show help pjsip show endpoints
No such command ‘pjsip show endpoints’.
localhost
CLI>
localhostCLI> module show
Module Description Use Count Status Support Level
acl Named ACL system 2 Running core
app_adsiprog.so Asterisk ADSI Programming Application 0 Running deprecated
app_agent_pool.so Call center agent pool applications 0 Running core
app_alarmreceiver.so Alarm Receiver for Asterisk 0 Running extended
app_amd.so Answering Machine Detection Application 0 Running extended
app_attended_transfer.so Attended transfer to the given extension 0 Running core
app_authenticate.so Authentication Application 0 Running core
app_blind_transfer.so Blind transfer channel to the given dest 0 Running core
app_bridgeaddchan.so Bridge Add Channel Application 0 Running core
app_bridgewait.so Place the channel into a holding bridge 0 Running core
app_cdr.so Tell Asterisk to not maintain a CDR for 0 Running core
app_celgenuserevent.so Generate an User-Defined CEL event 0 Running core
app_chanisavail.so Check channel availability 0 Running extended
app_channelredirect.so Redirects a given channel to a dialplan 0 Running core
app_chanspy.so Listen to the audio of an active channel 0 Running core
app_confbridge.so Conference Bridge Application 1 Running core
app_controlplayback.so Control Playback Application 0 Running core
app_dahdiras.so DAHDI ISDN Remote Access Server 0 Running deprecated
app_db.so Database Access Functions 0 Running core
app_dial.so Dialing Application 0 Running core
app_dictate.so Virtual Dictation Machine 0 Running extended
app_directed_pickup.so Directed Call Pickup Application 0 Running core
app_directory.so Extension Directory 0 Running core
app_disa.so DISA (Direct Inward System Access) Appli 0 Running core
app_dumpchan.so Dump Info About The Calling Channel 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_exec.so Executes dialplan applications 0 Running core
app_externalivr.so External IVR Interface Application 0 Running extended
app_festival.so Simple Festival Interface 0 Running extended
app_flash.so Flash channel application 0 Running core
app_followme.so Find-Me/Follow-Me Application 0 Running core
app_forkcdr.so Fork The CDR into 2 separate entities 0 Running core
app_getcpeid.so Get ADSI CPE ID 0 Running deprecated
app_ices.so Encode and Stream via icecast and ices 0 Running deprecated
app_image.so Image Transmission Application 0 Running deprecated
app_ivrdemo.so IVR Demo Application 0 Running extended
app_macro.so Extension Macros 0 Running deprecated
app_meetme.so MeetMe conference bridge 0 Running extended
app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 Running core
app_minivm.so Mini VoiceMail (A minimal Voicemail e-ma 0 Running extended
app_mixmonitor.so Mixed Audio Monitoring Application 0 Running core
app_morsecode.so Morse code 0 Running extended
app_mp3.so Silly MP3 Application 0 Running extended
app_nbscat.so Silly NBS Stream Application 0 Running deprecated
app_originate.so Originate call 0 Running core
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 0 Running core
app_playtones.so Playtones Application 0 Running core
app_privacy.so Require phone number to be entered, if n 0 Running core
app_queue.so True Call Queueing 0 Running core
app_read.so Read Variable Application 0 Running core
app_readexten.so Read and evaluate extension validity 0 Running core
app_record.so Trivial Record Application 0 Running core
app_saycounted.so Decline words according to channel langu 0 Running extended
app_sayunixtime.so Say time 0 Running core
app_senddtmf.so Send DTMF digits Application 0 Running core
app_sendtext.so Send Text Applications 0 Running core
app_skel.so Skeleton (sample) Application 0 Running core
app_sms.so SMS/PSTN handler 0 Running extended
app_softhangup.so Hangs up the requested channel 0 Running core
app_speech_utils.so Dialplan Speech Applications 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_stasis.so Stasis dialplan application 0 Running core
app_statsd.so StatsD Dialplan Application 0 Running extended
app_stream_echo.so Stream Echo Application 0 Running core
app_system.so Generic System() application 0 Running core
app_talkdetect.so Playback with Talk Detection 0 Running extended
app_test.so Interface Test Application 0 Running extended
app_transfer.so Transfers a caller to another extension 0 Running core
app_url.so Send URL Applications 0 Running deprecated
app_userevent.so Custom User Event Application 0 Running core
app_verbose.so Send verbose output 0 Running core
app_voicemail.so Comedian Mail (Voicemail System) 0 Not Running core
app_waitforring.so Waits until first ring after time 0 Running extended
app_waitforsilence.so Wait For Silence/Noise 0 Running extended
app_waituntil.so Wait until specified time 0 Running core
app_while.so While Loops and Conditional Execution 0 Running core
app_zapateller.so Block Telemarketers with Special Informa 0 Running extended
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_builtin_interval_features.so Built in bridging interval features 0 Running core
bridge_holding.so Holding bridge module 0 Running core
bridge_native_rtp.so Native RTP bridging module 0 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 0 Running core
ccss Call Completion Supplementary Services 4 Running core
cdr CDR Engine 8 Running core
cdr_csv.so Comma Separated Values CDR Backend 0 Running extended
cdr_custom.so Customizable Comma Separated Values CDR 0 Running core
cdr_manager.so Asterisk Manager Interface CDR Backend 0 Running core
cdr_sqlite3_custom.so SQLite3 Custom CDR Module 0 Not Running extended
cdr_syslog.so Customizable syslog CDR Backend 0 Not Running core
cel CEL Engine 5 Running core
cel_custom.so Customizable Comma Separated Values CEL 0 Running core
cel_manager.so Asterisk Manager Interface CEL Backend 0 Running core
cel_sqlite3_custom.so SQLite3 Custom CEL Module 0 Not Running extended
chan_bridge_media.so Bridge Media Channel Driver 0 Running core
chan_dahdi.so DAHDI Telephony w/PRI 0 Running core
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 Running core
chan_mgcp.so Media Gateway Control Protocol (MGCP) 0 Running extended
chan_ooh323.so Objective Systems H323 Channel 0 Running extended
chan_oss.so OSS Console Channel Driver 0 Running deprecated
chan_phone.so Linux Telephony API Support 0 Running deprecated
chan_rtp.so RTP Media Channel 0 Running core
chan_skinny.so Skinny Client Control Protocol (Skinny) 0 Running extended
chan_unistim.so UNISTIM Protocol (USTM) 0 Running extended
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 Running core
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 Running core
codec_alaw.so A-law Coder/Decoder 0 Running core
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 Running core
codec_g722.so ITU G.722-64kbps G722 Transcoder 0 Running core
codec_g726.so ITU G.726-32kbps G726 Transcoder 0 Running core
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_ilbc.so iLBC Coder/Decoder 0 Running core
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 Running core
codec_resample.so SLIN Resampling Codec 0 Running core
codec_ulaw.so mu-Law Coder/Decoder 0 Running core
dnsmgr DNS Manager 2 Running core
dsp DSP 1 Running core
enum ENUM Support 2 Running core
extconfig Configuration 14 Running core
features Call Features 1 Running core
format_g719.so ITU G.719 0 Running core
format_g723.so G.723.1 Simple Timestamp File Format 0 Running core
format_g726.so Raw G.726 (16/24/32/40kbps) data 0 Running core
format_g729.so Raw G.729 data 0 Running core
format_gsm.so Raw GSM data 0 Running core
format_h263.so Raw H.263 data 0 Running core
format_h264.so Raw H.264 data 0 Running core
format_ilbc.so Raw iLBC data 0 Running core
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 Running core
format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 Running core
format_sln.so Raw Signed Linear Audio support (SLN) 8k 0 Running core
format_vox.so Dialogic VOX (ADPCM) File Format 0 Running extended
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 Running core
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 Running core
func_aes.so AES dialplan functions 0 Running core
func_base64.so base64 encode/decode dialplan functions 0 Running core
func_blacklist.so Look up Caller
ID name/number from black 0 Running core
func_callcompletion.so Call Control Configuration Function 0 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_cdr.so Call Detail Record (CDR) dialplan functi 1 Running core
func_channel.so Channel information dialplan functions 0 Running core
func_config.so Asterisk configuration file variable acc 0 Running core
func_cut.so Cut out information from a string 0 Running core
func_db.so Database (astdb) related dialplan functi 0 Running core
func_devstate.so Gets or sets a device state in the dialp 0 Running core
func_dialgroup.so Dialgroup dialplan function 0 Running core
func_dialplan.so Dialplan Context/Extension/Priority Chec 0 Running core
func_enum.so ENUM related dialplan functions 0 Running core
func_env.so Environment/filesystem dialplan function 0 Running core
func_extstate.so Gets an extension’s state in the dialpla 0 Running core
func_frame_trace.so Frame Trace for internal ast_frame debug 0 Running extended
func_global.so Variable dialplan functions 0 Running core
func_groupcount.so Channel group dialplan functions 0 Running core
func_hangupcause.so HANGUPCAUSE related functions and applic 0 Running core
func_holdintercept.so Hold interception dialplan function 0 Running core
func_iconv.so Charset conversions 0 Running core
func_jitterbuffer.so Jitter buffer for read side of channel. 1 Running core
func_lock.so Dialplan mutexes 0 Running core
func_logic.so Logical dialplan functions 0 Running core
func_math.so Mathematical dialplan function 0 Running core
func_md5.so MD5 digest dialplan functions 0 Running core
func_module.so Checks if Asterisk module is loaded in m 0 Running core
func_periodic_hook.so Periodic dialplan hooks. 2 Running core
func_pitchshift.so Audio Effects Dialplan Functions 0 Running extended
func_presencestate.so Gets or sets a presence state in the dia 0 Running core
func_rand.so Random number dialplan function 0 Running core
func_realtime.so Read/Write/Store/Destroy values from a R 0 Running core
func_sha1.so SHA-1 computation dialplan function 0 Running core
func_shell.so Collects the output generated by a comma 0 Running core
func_sorcery.so Get a field from a sorcery object 0 Running core
func_sprintf.so SPRINTF dialplan function 0 Running core
func_srv.so SRV related dialplan functions 0 Running core
func_strings.so String handling dialplan functions 0 Running core
func_sysinfo.so System information related functions 0 Running core
func_talkdetect.so Talk detection dialplan function 0 Running core
func_timeout.so Channel timeout dialplan functions 0 Running core
func_uri.so URI encode/decode dialplan functions 0 Running core
func_version.so Get Asterisk Version/Build Info 0 Running core
func_vmcount.so Indicator for whether a voice mailbox ha 0 Running core
func_volume.so Technology independent volume control 0 Running core
http Built-in HTTP Server 5 Running core
indications Indication Tone Handling 1 Running core
logger Logger 1 Running core
manager Asterisk Manager Interface 1 Running core
pbx_ael.so Asterisk Extension Language Compiler 0 Running extended
pbx_config.so Text Extension Configuration 0 Running core
pbx_dundi.so Distributed Universal Number Discovery ( 0 Running extended
pbx_loopback.so Loopback Switch 0 Running core
pbx_realtime.so Realtime Switch 0 Running extended
pbx_spool.so Outgoing Spool Support 0 Running core
plc PLC 1 Running core
res_adsi.so ADSI Resource 3 Running deprecated
res_ael_share.so share-able code for AEL 1 Running extended
res_agi.so Asterisk Gateway Interface (AGI) 1 Running core
res_ari.so Asterisk RESTful Interface 10 Running core
res_ari_applications.so RESTful API module - Stasis application 0 Running core
res_ari_asterisk.so RESTful API module - Asterisk resources 0 Running core
res_ari_bridges.so RESTful API module - Bridge resources 0 Running core
res_ari_channels.so RESTful API module - Channel resources 0 Running core
res_ari_device_states.so RESTful API module - Device state resour 0 Running core
res_ari_endpoints.so RESTful API module - Endpoint resources 0 Running core
res_ari_events.so RESTful API module - WebSocket resource 0 Running core
res_ari_model.so ARI Model validators 10 Running core
res_ari_playbacks.so RESTful API module - Playback control re 0 Running core
res_ari_recordings.so RESTful API module - Recording resources 0 Running core
res_ari_sounds.so RESTful API module - Sound resources 0 Running core
res_calendar.so Asterisk Calendar integration 0 Running extended
res_chan_stats.so Example of how to use Stasis 0 Running extended
res_clialiases.so CLI Aliases 0 Running core
res_clioriginate.so Call origination and redirection from th 0 Running core
res_config_sqlite3.so SQLite 3 realtime config engine 0 Running core
res_convert.so File format conversion CLI command 0 Running core
res_crypto.so Cryptographic Digital Signatures 3 Running core
res_endpoint_stats.so Endpoint statistics 0 Running extended
res_fax.so Generic FAX Applications 0 Running core
res_format_attr_celt.so CELT Format Attribute Module 1 Running core
res_format_attr_g729.so G.729 Format Attribute Module 1 Running core
res_format_attr_h263.so H.263 Format Attribute Module 1 Running core
res_format_attr_h264.so H.264 Format Attribute Module 1 Running core
res_format_attr_ilbc.so iLBC Format Attribute Module 1 Running core
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
res_format_attr_silk.so SILK Format Attribute Module 1 Running core
res_format_attr_siren14.so Siren14 Format Attribute Module 1 Running core
res_format_attr_siren7.so Siren7 Format Attribute Module 1 Running core
res_format_attr_vp8.so VP8 Format Attribute Module 1 Running core
res_http_websocket.so HTTP WebSocket Support 2 Running extended
res_limit.so Resource limits 0 Running core
res_manager_devicestate.so Manager Device State Topic Forwarder 0 Running core
res_manager_presencestate.so Manager Presence State Topic Forwarder 0 Running core
res_monitor.so Call Monitoring Resource 2 Running deprecated
res_musiconhold.so Music On Hold Resource 0 Running core
res_mutestream.so Mute audio stream resources 0 Running core
res_mwi_devstate.so MWI Device State Subscriptions 0 Running core
res_mwi_external.so Core external MWI resource 1 Running core
res_mwi_external_ami.so AMI support for external MWI 0 Running core
res_parking.so Call Parking Resource 0 Running core
res_phoneprov.so HTTP Phone Provisioning 0 Running extended
res_pjproject.so PJPROJECT Log and Utility Support 1 Running core
res_pktccops.so PktcCOPS manager for MGCP 1 Running extended
res_realtime.so Realtime Data Lookup/Rewrite 0 Running core
res_remb_modifier.so REMB Modifier Module 0 Running extended
res_rtp_asterisk.so Asterisk RTP Stack 0 Running core
res_rtp_multicast.so Multicast RTP Engine 1 Running core
res_security_log.so Security Event Logging 0 Running core
res_smdi.so Simplified Message Desk Interface (SMDI) 2 Running extended
res_sorcery_astdb.so Sorcery Astdb Object Wizard 1 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 1 Running core
res_sorcery_memory.so Sorcery In-Memory Object Wizard 0 Running core
res_sorcery_memory_cache.so Sorcery Memory Cache Object Wizard 0 Running core
res_sorcery_realtime.so Sorcery Realtime Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 2 Running core
res_stasis.so Stasis application support 17 Running core
res_stasis_answer.so Stasis application answer support 1 Running core
res_stasis_device_state.so Stasis application device state support 1 Running core
res_stasis_playback.so Stasis application playback support 3 Running core
res_stasis_recording.so Stasis application recording support 4 Running core
res_stasis_snoop.so Stasis application snoop support 1 Running core
res_statsd.so StatsD client support 3 Running extended
res_stun_monitor.so STUN Network Monitor 0 Running core
res_timing_dahdi.so DAHDI Timing Interface 0 Running core
res_timing_pthread.so pthread Timing Interface 0 Running extended
res_timing_timerfd.so Timerfd Timing Interface 1 Running core
sounds Sounds Index 1 Running core
udptl UDPTL 2 Running core
266 modules loaded
localhost*CLI>

please suggest any solution for this, Iam verymuch thankful.

Regards,
Ramana

Server details:
RED HAT ENTERPRISE LINUX (RHEL) 7.8
Kernel Version 3.10.0-1127.10.1.el7.x86_64

It appears like you do not have either chan_sip or res_pjsip (and friends) loaded.

I suspect this is a new install so you should be using res_pjsip instead of chan_sip unless you have a reason to do otherwise.

Did you compile it?

If so, you may be able to locate it using:

'sudo find / -xdev -name res_pjsip.so'

If you do not find it, you need to compile it. If you do find it, you need to find out why it was not loaded. (Hint – look at modules.conf.)

I suspect you did not compile it.

Thank you very much sir for your kind response,

As suspected, this is a new installation of Asterisk in DELL server populated with degium analog and digital cards and RHEL 7.8

Accordinglybelow is the results observed, please look into this and suggest the .

[root@localhost ~]# sudo find / -xdev -name res_pjsip.so
/root/asterisk-17.4.0/res/res_pjsip.so
/usr/lib/asterisk/modules/res_pjsip.so
[root@localhost ~]#

and

etc/asterisk/modules.conf
;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger initialization) can be loaded
; using ‘preload’. ‘preload’ forces a module and the modules it
; is known to depend upon to be loaded earlier than they normally get
; loaded.
;
; NOTE: There is no good reason left to use ‘preload’ anymore. It was
; historically required to preload realtime driver modules so you could
; map Asterisk core configuration files to Realtime storage.
; This is no longer needed.
;
;preload => your_special_module.so
;
; If you want Asterisk to fail if a module does not load, then use
; the “require” keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
;
; require = chan_pjsip.so
;
; If you want you can combine with preload
; preload-require = your_special_module.so
;
;load => res_musiconhold.so
;
; Load one of: chan_oss, alsa, or console (portaudio).
; By default, load chan_oss only (automatically).
;
noload => chan_alsa.so
;noload => chan_oss.so
noload => chan_console.so

noload => res_hep.so
noload => res_hep_pjsip.so
noload => res_hep_rtcp.so

; Do not load chan_sip by default, it may conflict with res_pjsip.
noload => chan_sip.so

; The default voicemail module is app_voicemal. All voicemail modules
; are mutually exclusive. Therefore it is better to make sure they
; are not loaded at startup
;
noload => app_voicemail_odbc.so
noload => app_voicemail_imap.so

Try module load res_pjsip on the CLI. The error messages should indicate why it is not loading.

The reason that sip show peers fails is that you have explicitly configured chan_sip not to load.

Thank you very much sir for your kind response,

verified as suggested, below is the response please,

localhostCLI> module load res_pjsip.so
Unable to load module res_pjsip.so
Command ‘module load res_pjsip.so’ failed.
[Jul 17 19:22:12] WARNING[4585]: loader.c:1769 load_resource: Module ‘res_pjsip.so’ already loaded and running.
localhost
CLI>

As you can see the module is loaded. Try restarting Asterisk and then run your PJSIP commands

Which is a change from when the module list was generated!

We need a stable configuration to debug. I asked for the load command because the previous information showed the load had failed, and rerunning it should have produced the error messages associated with the failure. If if has loaded, one would expect the pjsip commands to have started to work.

Iam very much Thankful to you sir, and thanks to all,

Actually I have no knowledge on how to register SIP phone in asterisk. Depending on internet Iam trying hard to register Sip based Grandstream Ip phones (GXV1625) in Asterisk, but unsuccessful still now.

Can anyone please suggest that in which asterisk files I have to do configuration, and please share any sample configuration to register Grandstream sip based ip phones (numbers 7001, 7002 7003) in Asterisk (pjsip). Server is connected in 192.168.1.1 network.

[root@localhost ~]# ifconfig
em4: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500
inet 192.168.1.75 netmask 255.255.255.0 broadcast 192.168.1.255
inet6 fe80::fc93:8a38:3f5e:5813 prefixlen 64 scopeid 0x20
ether 80:18:44:ea:5a:cb txqueuelen 1000 (Ethernet)
RX packets 8735 bytes 700907 (684.4 KiB)
RX errors 0 dropped 12 overruns 0 frame 0
TX packets 5100 bytes 439035 (428.7 KiB)
TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0
device interrupt 79

With Regards
G.v.ramana

Assuming that the information you haven’t provided doesn’t include any surprises, see https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L297

However, the best way of getting support is to show what you have tried and provide logging showing how the registration was rejected. If you cannot get logging of the registration request, nothing you do in Asterisk can make a difference, as the request hasn’t got that far.

Dear all

Iam workion sip phones with Asterisk server.
3 no’s of Grandstream Ip phones registered succesfully by configuring pjsip.conf.
but incoming and outgoing calls are not working.
( when dialing number 600X error message observed in phone display as call failed: NO RESPONCE!).

Below is the Configuration details as I done.

etc/asterisk/pjsip.conf
;===============TRANSPORT

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============ENDPOINT TEMPLATES

endpoint-basic
type=endpoint
context=internal
disallow=all
allow=ulaw

auth-userpass
type=auth
auth_type=userpass

aor-single-reg
type=aor
max_contacts=1

;===============EXTENSION 6001

6001
auth=auth6001
aors=6001

auth6001
password=6001
username=6001

6001

;===============EXTENSION 6002

6002
auth=auth6002
aors=6002

auth6002
password=6002
username=6002

6002

;===============EXTENSION 6003

6003
auth=auth6003
aors=6003

auth6003
password=6003
username=6003

6003

etc/asterisk/extension.conf

[internal]
exten=>6001,1,Dial(PJSIP/6001,20)
exten=>6002,1,Dial(PJSIP/6002,20)
exten=>6003,1,Dial(PJSIP/6003,20)

asterisk*CLI> pjsip show endpoints

Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <criteria…>
Channel: <ChannelId…> <State…> <Time…>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>

Endpoint: 6001 Not in use 0 of inf
InAuth: auth6001/6001
Aor: 6001 1
Contact: 6001/sip:6001@192.168.1.14:5060 30ba26a28b NonQual nan

Endpoint: 6002 Not in use 0 of inf
InAuth: auth6002/6002
Aor: 6002 1
Contact: 6002/sip:6002@192.168.1.13:5060 236e107934 NonQual nan

Endpoint: 6003 Not in use 0 of inf
InAuth: auth6003/6003
Aor: 6003 1
Contact: 6003/sip:6003@192.168.1.12:5060 77df9397d9 NonQual nan

Objects found: 3

Requesting please suggest to work calls.

Thanks and Regards
Ramana

Hello,

you have to paste here what appear on Asterisk Console when you try to call another extension/endpoint.

Regards

Dear Sir,
Thank you very much for your response,

There is no any message or error on Asterisk console when I try to call another extension/endpoint.

Sir, can please correct the that dial-plan written in extension.conf Iam not sure it is correct or not.

Thanks and Regards
Ramana

Calls are working when I am dialing IP Address of the another endpoint.
but calls not working when dialing extension number.
In both cases asterisk CLI not showing any text/message.

requesting all, please suggest any solution for this regard.

Thanks and Regards
Ramana