Sip not wotking in Asterisk server

Asterisk Asterisk SIP
1

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)
localhostCLI>

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

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
localhostCLI>

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

localhostCLI> core show help pjsip show endpoints
No such command ‘pjsip show endpoints’.
localhostCLI>
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 CallerID 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

2

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

3

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.

4

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

5

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.

6

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.
localhostCLI>

7

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

8

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.

9

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

10

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.

11

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

12

Hello,

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

Regards

13

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

14

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

15

Thanks to all,

Phones are registered and calls are also working fine, by changing the configuration in pjsip.confand extension.conf

Regards
ramana

16

Dear all,

I got registered Video and audio phone’s in Asterisk server.
Incoming and outgoing calls are working fine.

There is one concern when dialing an extension number some delay (around 10 to 15 seconds) is observed to connect call. Except that everything working fine.

Here 6001 is one of the Video phone configuration and 6004 is one of the audio phone configuration.

Also please find the message history in Asterisk CLI>

Requesting all please support in this regard.

Thanks and Regards
Ramana

pjsip.conf

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

[6001]
type = endpoint
context = internal
disallow = all
allow = h264
allow = ulaw
aors = 6001
auth = auth6001

[6001]
type = aor
max_contacts = 1

[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6004]
type = endpoint
context = internal
disallow = all
allow = ulaw
aors = 6004
auth = auth6004

[6004]
type = aor
max_contacts = 1

[auth6004]
type=auth
auth_type=userpass
password=6004
username=6004

extensions.conf

[internal]
exten=>6001,1,Dial(PJSIP/6001)
exten=>6004,1,Dial(PJSIP/6004)

Traffic mesage History:

asteriskCLI>
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.75’
– Executing [6002@internal:1] Dial(“PJSIP/6001-0000000e”, “PJSIP/6002”) in new stack
– Called PJSIP/6002
– PJSIP/6002-0000000f is ringing
– PJSIP/6002-0000000f is ringing
– PJSIP/6002-0000000f answered PJSIP/6001-0000000e
– Channel PJSIP/6002-0000000f joined ‘simple_bridge’ basic-bridge <1c5852db-7286-44d9-b3cc-e6a0dfe27901>
– Channel PJSIP/6001-0000000e joined ‘simple_bridge’ basic-bridge <1c5852db-7286-44d9-b3cc-e6a0dfe27901>
– Channel PJSIP/6001-0000000e left ‘simple_bridge’ basic-bridge <1c5852db-7286-44d9-b3cc-e6a0dfe27901>
– Channel PJSIP/6002-0000000f left ‘simple_bridge’ basic-bridge <1c5852db-7286-44d9-b3cc-e6a0dfe27901>
== Spawn extension (internal, 6002, 1) exited non-zero on ‘PJSIP/6001-0000000e’
asteriskCLI>
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.75’
– Executing [6003@internal:1] Dial(“PJSIP/6002-00000010”, “PJSIP/6003”) in new stack
– Called PJSIP/6003
– PJSIP/6003-00000011 is ringing
– PJSIP/6003-00000011 is ringing
– PJSIP/6003-00000011 answered PJSIP/6002-00000010
– Channel PJSIP/6003-00000011 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/6002-00000010 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/6002-00000010 left ‘simple_bridge’ basic-bridge
– Channel PJSIP/6003-00000011 left ‘simple_bridge’ basic-bridge
== Spawn extension (internal, 6003, 1) exited non-zero on ‘PJSIP/6002-00000010’
asteriskCLI>
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.75’
– Executing [6004@internal:1] Dial(“PJSIP/6001-00000012”, “PJSIP/6004”) in new stack
– Called PJSIP/6004
– PJSIP/6004-00000013 is ringing
– PJSIP/6004-00000013 is ringing
== Spawn extension (internal, 6004, 1) exited non-zero on ‘PJSIP/6001-00000012’
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.1.75’
– Executing [6005@internal:1] Dial(“PJSIP/6002-00000014”, “PJSIP/6005”) in new stack
– Called PJSIP/6005
– PJSIP/6005-00000015 is ringing
– PJSIP/6005-00000015 is ringing
– PJSIP/6005-00000015 answered PJSIP/6002-00000014
– Channel PJSIP/6005-00000015 joined ‘simple_bridge’ basic-bridge <25e35767-5194-4512-a59c-a15aa89e069c>
– Channel PJSIP/6002-00000014 joined ‘simple_bridge’ basic-bridge <25e35767-5194-4512-a59c-a15aa89e069c>
– Channel PJSIP/6002-00000014 left ‘native_rtp’ basic-bridge <25e35767-5194-4512-a59c-a15aa89e069c>
– Channel PJSIP/6005-00000015 left ‘native_rtp’ basic-bridge <25e35767-5194-4512-a59c-a15aa89e069c>
== Spawn extension (internal, 6005, 1) exited non-zero on ‘PJSIP/6002-00000014’
asteriskCLI>

pjsip show history

No. Timestamp (Dir) Address SIP Message
===== ========== ============================== ===================================
00000 1596716228 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00001 1596716228 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00002 1596716228 * <== 192.168.1.11:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00003 1596716228 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00004 1596716228 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00005 1596716228 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00006 1596716228 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00007 1596716228 * <== 192.168.1.12:5060 SIP/2.0 180 Ringing
00008 1596716228 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00009 1596716232 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00010 1596716232 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00011 1596716232 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00012 1596716232 * <== 192.168.1.11:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00013 1596716233 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00014 1596716233 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00015 1596716233 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00016 1596716233 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00017 1596716233 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00018 1596716233 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00019 1596716233 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00020 1596716233 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00021 1596716233 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00022 1596716233 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00023 1596716233 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00024 1596716233 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00025 1596716233 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00026 1596716233 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00027 1596716233 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00028 1596716233 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00029 1596716233 * <== 192.168.1.12:5060 BYE sip:asterisk@192.168.1.75:5060 SIP/2.0
00030 1596716233 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00031 1596716233 * ==> 192.168.1.11:5060 BYE sip:6001@192.168.1.11:5060 SIP/2.0
00032 1596716233 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00033 1596716240 * <== 192.168.1.11:5060 INVITE sip:6003@192.168.1.75 SIP/2.0
00034 1596716240 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00035 1596716240 * <== 192.168.1.11:5060 ACK sip:6003@192.168.1.75 SIP/2.0
00036 1596716240 * <== 192.168.1.11:5060 INVITE sip:6003@192.168.1.75 SIP/2.0
00037 1596716240 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00038 1596716240 * ==> 192.168.1.13:5060 INVITE sip:6003@192.168.1.13:5060 SIP/2.0
00039 1596716240 * <== 192.168.1.13:5060 SIP/2.0 100 Trying
00040 1596716240 * <== 192.168.1.13:5060 SIP/2.0 180 Ringing
00041 1596716240 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00042 1596716243 * <== 192.168.1.11:5060 CANCEL sip:6003@192.168.1.75 SIP/2.0
00043 1596716243 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00044 1596716243 * ==> 192.168.1.11:5060 SIP/2.0 487 Request Terminated
00045 1596716243 * ==> 192.168.1.13:5060 CANCEL sip:6003@192.168.1.13:5060 SIP/2.0
00046 1596716243 * <== 192.168.1.11:5060 ACK sip:6003@192.168.1.75 SIP/2.0
00047 1596716243 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00048 1596716243 * <== 192.168.1.13:5060 SIP/2.0 487 Request Terminated
00049 1596716243 * ==> 192.168.1.13:5060 ACK sip:6003@192.168.1.13:5060 SIP/2.0
00050 1596716256 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00051 1596716256 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00052 1596716256 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00053 1596716256 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00054 1596716256 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00055 1596716256 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00056 1596716257 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00057 1596716258 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00058 1596716260 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00059 1596716264 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00060 1596716272 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00061 1596716288 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00062 1596716288 * ==> 192.168.1.11:5060 SIP/2.0 503 Service Unavailable
00063 1596716288 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00064 1596716697 * <== 192.168.2.16:5060 REGISTER sip:192.168.1.75 SIP/2.0
00065 1596716697 * ==> 192.168.2.16:5060 SIP/2.0 401 Unauthorized
00066 1596716697 * <== 192.168.2.16:5060 REGISTER sip:192.168.1.75 SIP/2.0
00067 1596716697 * ==> 192.168.2.16:5060 SIP/2.0 200 OK
00068 1596716701 * <== 192.168.2.15:5060 REGISTER sip:192.168.1.75 SIP/2.0
00069 1596716701 * ==> 192.168.2.15:5060 SIP/2.0 401 Unauthorized
00070 1596716702 * <== 192.168.2.15:5060 REGISTER sip:192.168.1.75 SIP/2.0
00071 1596716702 * ==> 192.168.2.15:5060 SIP/2.0 200 OK
00072 1596716707 * <== 192.168.2.14:5060 REGISTER sip:192.168.1.75 SIP/2.0
00073 1596716707 * ==> 192.168.2.14:5060 SIP/2.0 401 Unauthorized
00074 1596716707 * <== 192.168.2.14:5060 REGISTER sip:192.168.1.75 SIP/2.0
00075 1596716707 * ==> 192.168.2.14:5060 SIP/2.0 200 OK
00076 1596716949 * <== 192.168.1.11:5060 INVITE sip:6005@192.168.1.75 SIP/2.0
00077 1596716949 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00078 1596716949 * <== 192.168.1.11:5060 ACK sip:6005@192.168.1.75 SIP/2.0
00079 1596716949 * <== 192.168.1.11:5060 INVITE sip:6005@192.168.1.75 SIP/2.0
00080 1596716949 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00081 1596716964 * ==> 192.168.2.15:5060 INVITE sip:6005@192.168.2.15:5060 SIP/2.0
00082 1596716964 * <== 192.168.2.15:5060 SIP/2.0 100 Trying
00083 1596716964 * <== 192.168.2.15:5060 SIP/2.0 180 Ringing
00084 1596716964 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00085 1596716966 * <== 192.168.1.11:5060 CANCEL sip:6005@192.168.1.75 SIP/2.0
00086 1596716966 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00087 1596716966 * ==> 192.168.1.11:5060 SIP/2.0 487 Request Terminated
00088 1596716966 * ==> 192.168.2.15:5060 CANCEL sip:6005@192.168.2.15:5060 SIP/2.0
00089 1596716966 * <== 192.168.1.11:5060 ACK sip:6005@192.168.1.75 SIP/2.0
00090 1596716966 * <== 192.168.2.15:5060 SIP/2.0 200 OK
00091 1596716966 * <== 192.168.2.15:5060 SIP/2.0 487 Request Terminated
00092 1596716966 * ==> 192.168.2.15:5060 ACK sip:6005@192.168.2.15:5060 SIP/2.0
00093 1596716980 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00094 1596716980 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00095 1596716980 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00096 1596716980 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00097 1596716980 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00098 1596716990 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00099 1596716990 * <== 192.168.2.14:5060 SIP/2.0 100 Trying
00100 1596716990 * <== 192.168.2.14:5060 SIP/2.0 180 Ringing
00101 1596716990 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00102 1596716992 * <== 192.168.1.11:5060 CANCEL sip:6004@192.168.1.75 SIP/2.0
00103 1596716992 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00104 1596716992 * ==> 192.168.1.11:5060 SIP/2.0 487 Request Terminated
00105 1596716992 * ==> 192.168.2.14:5060 CANCEL sip:6004@192.168.2.14:5060 SIP/2.0
00106 1596716992 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00107 1596716992 * <== 192.168.2.14:5060 SIP/2.0 200 OK
00108 1596716992 * <== 192.168.2.14:5060 SIP/2.0 487 Request Terminated
00109 1596716992 * ==> 192.168.2.14:5060 ACK sip:6004@192.168.2.14:5060 SIP/2.0
00110 1596717060 * <== 192.168.1.11:5060 INVITE sip:6006@192.168.1.75 SIP/2.0
00111 1596717060 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00112 1596717060 * <== 192.168.1.11:5060 ACK sip:6006@192.168.1.75 SIP/2.0
00113 1596717060 * <== 192.168.1.11:5060 INVITE sip:6006@192.168.1.75 SIP/2.0
00114 1596717060 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00115 1596717070 * ==> 192.168.2.16:5060 INVITE sip:6006@192.168.2.16:5060 SIP/2.0
00116 1596717070 * <== 192.168.2.16:5060 SIP/2.0 100 Trying
00117 1596717070 * <== 192.168.2.16:5060 SIP/2.0 180 Ringing
00118 1596717070 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00119 1596717072 * <== 192.168.1.11:5060 CANCEL sip:6006@192.168.1.75 SIP/2.0
00120 1596717072 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00121 1596717072 * ==> 192.168.1.11:5060 SIP/2.0 487 Request Terminated
00122 1596717072 * ==> 192.168.2.16:5060 CANCEL sip:6006@192.168.2.16:5060 SIP/2.0
00123 1596717072 * <== 192.168.1.11:5060 ACK sip:6006@192.168.1.75 SIP/2.0
00124 1596717072 * <== 192.168.2.16:5060 SIP/2.0 200 OK
00125 1596717072 * <== 192.168.2.16:5060 SIP/2.0 487 Request Terminated
00126 1596717072 * ==> 192.168.2.16:5060 ACK sip:6006@192.168.2.16:5060 SIP/2.0
00127 1596717096 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00128 1596717096 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00129 1596717096 * <== 192.168.1.11:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00130 1596717096 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00131 1596717096 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00132 1596717116 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00133 1596717116 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00134 1596717116 * <== 192.168.1.12:5060 SIP/2.0 180 Ringing
00135 1596717116 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00136 1596717119 * <== 192.168.1.12:5060 SIP/2.0 486 Busy Here
00137 1596717119 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00138 1596717119 * ==> 192.168.1.11:5060 SIP/2.0 486 Busy Here
00139 1596717119 * <== 192.168.1.11:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00140 1596717142 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00141 1596717142 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00142 1596717142 * <== 192.168.1.11:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00143 1596717142 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00144 1596717142 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00145 1596717172 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00146 1596717172 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00147 1596717172 * <== 192.168.1.12:5060 SIP/2.0 180 Ringing
00148 1596717172 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00149 1596717175 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00150 1596717175 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00151 1596717175 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00152 1596717175 * <== 192.168.1.11:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00153 1596717176 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00154 1596717176 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00155 1596717176 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00156 1596717176 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00157 1596717176 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00158 1596717176 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00159 1596717176 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00160 1596717176 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00161 1596717176 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00162 1596717176 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00163 1596717176 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00164 1596717176 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00165 1596717177 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00166 1596717177 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00167 1596717177 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00168 1596717177 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00169 1596717177 * <== 192.168.1.11:5060 BYE sip:192.168.1.75:5060 SIP/2.0
00170 1596717177 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00171 1596717177 * ==> 192.168.1.12:5060 BYE sip:6002@192.168.1.12:5060 SIP/2.0
00172 1596717177 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00173 1596717200 * <== 192.168.1.12:5060 INVITE sip:6003@192.168.1.75 SIP/2.0
00174 1596717200 * ==> 192.168.1.12:5060 SIP/2.0 401 Unauthorized
00175 1596717200 * <== 192.168.1.12:5060 ACK sip:6003@192.168.1.75 SIP/2.0
00176 1596717200 * <== 192.168.1.12:5060 INVITE sip:6003@192.168.1.75 SIP/2.0
00177 1596717200 * ==> 192.168.1.12:5060 SIP/2.0 100 Trying
00178 1596717220 * ==> 192.168.1.13:5060 INVITE sip:6003@192.168.1.13:5060 SIP/2.0
00179 1596717220 * <== 192.168.1.13:5060 SIP/2.0 100 Trying
00180 1596717220 * <== 192.168.1.13:5060 SIP/2.0 180 Ringing
00181 1596717220 * ==> 192.168.1.12:5060 SIP/2.0 180 Ringing
00182 1596717223 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00183 1596717223 * ==> 192.168.1.13:5060 ACK sip:6003@192.168.1.13:5060 SIP/2.0
00184 1596717223 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00185 1596717223 * <== 192.168.1.12:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00186 1596717224 * <== 192.168.1.12:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00187 1596717224 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00188 1596717224 * ==> 192.168.1.13:5060 INFO sip:6003@192.168.1.13:5060 SIP/2.0
00189 1596717224 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00190 1596717224 * <== 192.168.1.13:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00191 1596717224 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00192 1596717224 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00193 1596717224 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00194 1596717224 * <== 192.168.1.13:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00195 1596717224 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00196 1596717224 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00197 1596717224 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00198 1596717224 * <== 192.168.1.12:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00199 1596717224 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00200 1596717224 * ==> 192.168.1.13:5060 INFO sip:6003@192.168.1.13:5060 SIP/2.0
00201 1596717224 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00202 1596717224 * <== 192.168.1.12:5060 BYE sip:192.168.1.75:5060 SIP/2.0
00203 1596717224 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00204 1596717224 * ==> 192.168.1.13:5060 BYE sip:6003@192.168.1.13:5060 SIP/2.0
00205 1596717224 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00206 1596717241 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00207 1596717241 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00208 1596717241 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00209 1596717241 * <== 192.168.1.11:5060 INVITE sip:6004@192.168.1.75 SIP/2.0
00210 1596717241 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00211 1596717251 * ==> 192.168.2.14:5060 INVITE sip:6004@192.168.2.14:5060 SIP/2.0
00212 1596717251 * <== 192.168.2.14:5060 SIP/2.0 100 Trying
00213 1596717251 * <== 192.168.2.14:5060 SIP/2.0 180 Ringing
00214 1596717251 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00215 1596717252 * <== 192.168.1.11:5060 CANCEL sip:6004@192.168.1.75 SIP/2.0
00216 1596717252 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00217 1596717252 * ==> 192.168.1.11:5060 SIP/2.0 487 Request Terminated
00218 1596717252 * ==> 192.168.2.14:5060 CANCEL sip:6004@192.168.2.14:5060 SIP/2.0
00219 1596717252 * <== 192.168.1.11:5060 ACK sip:6004@192.168.1.75 SIP/2.0
00220 1596717252 * <== 192.168.2.14:5060 SIP/2.0 200 OK
00221 1596717252 * <== 192.168.2.14:5060 SIP/2.0 487 Request Terminated
00222 1596717252 * ==> 192.168.2.14:5060 ACK sip:6004@192.168.2.14:5060 SIP/2.0
00223 1596717277 * <== 192.168.1.12:5060 INVITE sip:6005@192.168.1.75 SIP/2.0
00224 1596717277 * ==> 192.168.1.12:5060 SIP/2.0 401 Unauthorized
00225 1596717278 * <== 192.168.1.12:5060 ACK sip:6005@192.168.1.75 SIP/2.0
00226 1596717278 * <== 192.168.1.12:5060 INVITE sip:6005@192.168.1.75 SIP/2.0
00227 1596717278 * ==> 192.168.1.12:5060 SIP/2.0 100 Trying
00228 1596717288 * ==> 192.168.2.15:5060 INVITE sip:6005@192.168.2.15:5060 SIP/2.0
00229 1596717288 * <== 192.168.2.15:5060 SIP/2.0 100 Trying
00230 1596717288 * <== 192.168.2.15:5060 SIP/2.0 180 Ringing
00231 1596717288 * ==> 192.168.1.12:5060 SIP/2.0 180 Ringing
00232 1596717291 * <== 192.168.2.15:5060 SIP/2.0 200 OK
00233 1596717291 * ==> 192.168.2.15:5060 ACK sip:6005@192.168.2.15:5060 SIP/2.0
00234 1596717291 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00235 1596717291 * ==> 192.168.2.15:5060 INVITE sip:6005@192.168.2.15:5060 SIP/2.0
00236 1596717291 * <== 192.168.1.12:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00237 1596717291 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00238 1596717291 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00239 1596717291 * <== 192.168.2.15:5060 SIP/2.0 100 Trying
00240 1596717291 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00241 1596717291 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00242 1596717292 * <== 192.168.2.15:5060 SIP/2.0 200 OK
00243 1596717292 * ==> 192.168.2.15:5060 ACK sip:6005@192.168.2.15:5060 SIP/2.0
00244 1596717294 * <== 192.168.1.12:5060 BYE sip:192.168.1.75:5060 SIP/2.0
00245 1596717294 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00246 1596717294 * ==> 192.168.2.15:5060 INVITE sip:6005@192.168.2.15:5060 SIP/2.0
00247 1596717294 * <== 192.168.2.15:5060 SIP/2.0 100 Trying
00248 1596717294 * <== 192.168.2.15:5060 SIP/2.0 200 OK
00249 1596717294 * ==> 192.168.2.15:5060 ACK sip:6005@192.168.2.15:5060 SIP/2.0
00250 1596717294 * ==> 192.168.2.15:5060 BYE sip:6005@192.168.2.15:5060 SIP/2.0
00251 1596717294 * <== 192.168.2.15:5060 SIP/2.0 200 OK
00252 1596717800 * <== 192.168.1.13:5060 REGISTER sip:192.168.1.75 SIP/2.0
00253 1596717800 * ==> 192.168.1.13:5060 SIP/2.0 401 Unauthorized
00254 1596717800 * <== 192.168.1.13:5060 REGISTER sip:192.168.1.75 SIP/2.0
00255 1596717800 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00256 1596717802 * <== 192.168.1.11:5060 REGISTER sip:192.168.1.75 SIP/2.0
00257 1596717802 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00258 1596717802 * <== 192.168.1.11:5060 REGISTER sip:192.168.1.75 SIP/2.0
00259 1596717802 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00260 1596717804 * <== 192.168.1.12:5060 REGISTER sip:192.168.1.75 SIP/2.0
00261 1596717804 * ==> 192.168.1.12:5060 SIP/2.0 401 Unauthorized
00262 1596717804 * <== 192.168.1.12:5060 REGISTER sip:192.168.1.75 SIP/2.0
00263 1596717804 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00264 1596718332 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00265 1596718332 * ==> 192.168.1.11:5060 SIP/2.0 401 Unauthorized
00266 1596718332 * <== 192.168.1.11:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00267 1596718332 * <== 192.168.1.11:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00268 1596718332 * ==> 192.168.1.11:5060 SIP/2.0 100 Trying
00269 1596718352 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00270 1596718352 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00271 1596718352 * <== 192.168.1.12:5060 SIP/2.0 180 Ringing
00272 1596718352 * ==> 192.168.1.11:5060 SIP/2.0 180 Ringing
00273 1596718355 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00274 1596718355 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00275 1596718355 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00276 1596718355 * <== 192.168.1.11:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00277 1596718356 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00278 1596718356 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00279 1596718356 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00280 1596718356 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00281 1596718356 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00282 1596718356 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00283 1596718356 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00284 1596718356 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00285 1596718356 * <== 192.168.1.11:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00286 1596718356 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00287 1596718356 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00288 1596718356 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00289 1596718357 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00290 1596718357 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00291 1596718357 * ==> 192.168.1.11:5060 INFO sip:6001@192.168.1.11:5060 SIP/2.0
00292 1596718357 * <== 192.168.1.11:5060 SIP/2.0 200 OK
00293 1596718362 * <== 192.168.1.11:5060 BYE sip:192.168.1.75:5060 SIP/2.0
00294 1596718362 * ==> 192.168.1.11:5060 SIP/2.0 200 OK
00295 1596718362 * ==> 192.168.1.12:5060 BYE sip:6002@192.168.1.12:5060 SIP/2.0
00296 1596718362 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00297 1596718397 * <== 192.168.1.13:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00298 1596718397 * ==> 192.168.1.13:5060 SIP/2.0 401 Unauthorized
00299 1596718397 * <== 192.168.1.13:5060 ACK sip:6002@192.168.1.75 SIP/2.0
00300 1596718397 * <== 192.168.1.13:5060 INVITE sip:6002@192.168.1.75 SIP/2.0
00301 1596718397 * ==> 192.168.1.13:5060 SIP/2.0 100 Trying
00302 1596718417 * ==> 192.168.1.12:5060 INVITE sip:6002@192.168.1.12:5060 SIP/2.0
00303 1596718417 * <== 192.168.1.12:5060 SIP/2.0 100 Trying
00304 1596718417 * <== 192.168.1.12:5060 SIP/2.0 180 Ringing
00305 1596718417 * ==> 192.168.1.13:5060 SIP/2.0 180 Ringing
00306 1596718419 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00307 1596718419 * ==> 192.168.1.12:5060 ACK sip:6002@192.168.1.12:5060 SIP/2.0
00308 1596718419 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00309 1596718419 * <== 192.168.1.13:5060 ACK sip:192.168.1.75:5060 SIP/2.0
00310 1596718421 * <== 192.168.1.13:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00311 1596718421 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00312 1596718421 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00313 1596718421 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00314 1596718421 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00315 1596718421 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00316 1596718421 * ==> 192.168.1.13:5060 INFO sip:6003@192.168.1.13:5060 SIP/2.0
00317 1596718421 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00318 1596718421 * <== 192.168.1.12:5060 INFO sip:asterisk@192.168.1.75:5060 SIP/2.0
00319 1596718421 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00320 1596718421 * ==> 192.168.1.13:5060 INFO sip:6003@192.168.1.13:5060 SIP/2.0
00321 1596718421 * <== 192.168.1.13:5060 SIP/2.0 200 OK
00322 1596718421 * <== 192.168.1.13:5060 INFO sip:192.168.1.75:5060 SIP/2.0
00323 1596718421 * ==> 192.168.1.13:5060 SIP/2.0 200 OK
00324 1596718421 * ==> 192.168.1.12:5060 INFO sip:6002@192.168.1.12:5060 SIP/2.0
00325 1596718421 * <== 192.168.1.12:5060 SIP/2.0 200 OK
00326 1596718422 * <== 192.168.1.12:5060 BYE sip:asterisk@192.168.1.75:5060 SIP/2.0
00327 1596718422 * ==> 192.168.1.12:5060 SIP/2.0 200 OK
00328 1596718422 * ==> 192.168.1.13:5060 BYE sip:6003@192.168.1.13:5060 SIP/2.0
00329 1596718422 * <== 192.168.1.13:5060 SIP/2.0 200 OK

17

That would normally indicate an incomplete domain name resolution configuration which is resulting in making DNS queries that time out.

19

Thank you very much sir,

I will look into that, and come back with results soon, from last few days Iam busy in other work.

Regards
Ramana

20

Thanks to all,

Delay when dailing pjsip extensions is reduced considerably after making changes in OSPF routing of network devices connected to asterisk server. kept in observation on delay, at this point working well.

Regards
ramana

21

Dear all,

Is it is possible to make calls between PJSIP endpoints (SIP Phones) and Analog phones connected to Degium 8 port FXS module (A8B06F), Please suggest if there is an any option.

Thanks and regards
Ramana