Etisalat Trunk SIP Line

[abdulnawas]

Hi,
Below is my configuration for Carriers for my sip trunk line.

register => 42530300.etisalat:Dummy@192.168.1.6:5060/42530300

Account Entry:
[42530300]
type=peer
username=42530300
secret=Dummy
fromuser=42530300
host=192.168.1.6
dtmfmode=rfc2833
context==default
insecure=very
disallow=all
canreivite=no
qualify=300

Dialplan Entry:
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${SIPDXB}/${EXTEN:})
exten => _X.,3,Hangup

I am very much new to Asterisks and Vicidial. When i dial on the above number it will busy tone and i can't dial out, i get below answers
vicibox*CLI> console dial 0505474840
[Sep 15 07:48:15] WARNING[24707]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Sep 15 07:48:15] NOTICE[24707]: console_video.c:137 console_video_start: voice only, console video support not present
[Sep 15 07:48:15] -- Executing [0505474840@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 15 07:48:15] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 15 07:48:15] -- Executing [0505474840@default:2] Dial("Console/dsp", "SIP/42530300/0505474840") in new stack
[Sep 15 07:48:15] == Using SIP RTP CoS mark 5
[Sep 15 07:48:15] WARNING[35062]: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 0505474840
[Sep 15 07:48:15] -- Couldn't call SIP/42530300/0505474840
[Sep 15 07:48:15] == Everyone is busy/congested at this time (0:0/0/0)
[Sep 15 07:48:15] -- Executing [0505474840@default:3] Hangup("Console/dsp", "") in new stack
[Sep 15 07:48:15] == Spawn extension (default, 0505474840, 3) exited non-zero on 'Console/dsp'
[Sep 15 07:48:15] -- Executing [h@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Sep 15 07:48:15] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 15 07:48:15] << Hangup on console >>
vicibox*CLI>

Please help me to rectify this issue

Thank You and Regards

Abdul

[geoff3dmg]

your account entry is wrong. You said to 'disallow=all' but then did not allow any audio codecs (e.g. allow=ulaw).

[abdulnawas]

I have changed it please see the configuration below

[Etisalat]
disallow=all
allow=alaw
allow=ulaw
type=peer
username=42530300
secret=Dummy
host=192.168.1.6
dtmfmode=rfc2833
context=trunkinbound
outboundproxy=192.168.1.6
transport=UDP

still the same

[geoff3dmg]

do you still get this warning?

Code: Select all

[Sep 15 07:48:15] WARNING[35062]: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 0505474840

[abdulnawas]

Yes,
It was registered but i am not getting incoming call or i cannot dial out. but when i use with xlite client working great, my service provider given me below details
UserID:42530300
Password:Dummy
Domain Name: 42530300.etisalat
CEP IP= Public Address 192.168.1.11 and SBC IP Address 192.168.1.6
Vlan id = 200
LLCA = DUBAILLCA
Channel 30
Pilot number 42530300

[abdulnawas]

it get this

vicibox*CLI> sip reload
[Sep 16 13:46:21] Reloading SIP
[Sep 16 13:46:21] == Parsing '/etc/asterisk/sip.conf': [Sep 16 13:46:21] == Found
[Sep 16 13:46:21] == Parsing '/etc/asterisk/sip-vicidial.conf': [Sep 16 13:46:21] == Found
[Sep 16 13:46:21] == Parsing '/etc/asterisk/users.conf': [Sep 16 13:46:21] == Found
[Sep 16 13:46:21] > removing dns manager for '42530300.etisalat'
[Sep 16 13:46:21] == Using SIP CoS mark 4
[Sep 16 13:46:21] == Parsing '/etc/asterisk/sip_notify.conf': [Sep 16 13:46:21] == Found
[Sep 16 13:46:22] > doing dnsmgr_lookup for '42530300.etisalat'
[Sep 16 13:46:22] ERROR[2170]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("42530300.etisalat", "(null)", ...): Name or service not known
[Sep 16 13:46:22] WARNING[2170]: acl.c:590 resolve_first: Unable to lookup '42530300.etisalat'
[Sep 16 13:46:22] > adding dns manager for '42530300.etisalat'
[Sep 16 13:46:22] WARNING[2170]: acl.c:719 ast_ouraddrfor: Cannot connect
[Sep 16 13:46:22] WARNING[2170]: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x7fe458000d70 (len 401) to (null) returned -1: Invalid argument
[Sep 16 13:46:22] WARNING[2170]: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x7fe458000d70 (len 401) to (null) returned -1: Invalid argument
[Sep 16 13:46:23] WARNING[2170]: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x7fe458000d70 (len 401) to (null) returned -1: Invalid argument
[Sep 16 13:46:25] WARNING[2170]: chan_sip.c:3714 __sip_xmit: sip_xmit of 0x7fe458000d70 (len 401) to (null) returned -1: Invalid argument

[geoff3dmg]

You are telling it to conenct to 42530300.etisalat where as your VoIP provider stated the ip to connect to is 192.168.1.6

[abdulnawas]

Thank you for your support

now i get below warning message

[Sep 16 14:13:45] WARNING[2170]: chan_sip.c:21487 handle_response_register: Got 404 Not found on SIP register to service 42530300@192.168.1.6, giving up

[abdulnawas]

I have sip registration says ok,

vicibox*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
300/300 (Unspecified) D N 0 UNKNOWN
301/301 192.168.11.17 D N 42693 OK (2 ms)
302/302 192.168.11.17 D N 42693 OK (2 ms)
303/303 (Unspecified) D N 0 UNKNOWN
304/304 (Unspecified) D N 0 UNKNOWN
305/305 (Unspecified) D N 0 UNKNOWN
307/307 (Unspecified) D N 0 UNKNOWN
308/308 (Unspecified) D N 0 UNKNOWN
309/309 (Unspecified) D N 0 UNKNOWN
310/310 (Unspecified) D N 0 UNKNOWN
Etisalat/42530300 192.168.1.6 N 5060 OK (4 ms)
11 sip peers [Monitored: 3 online, 8 offline Unmonitored: 0 online, 0 offline]
vicibox*CLI>

vicibox*CLI> sip reload
[Sep 16 14:23:28] Reloading SIP
[Sep 16 14:23:28] == Parsing '/etc/asterisk/sip.conf': [Sep 16 14:23:28] == Found
[Sep 16 14:23:28] == Parsing '/etc/asterisk/sip-vicidial.conf': [Sep 16 14:23:28] == Found
[Sep 16 14:23:28] == Parsing '/etc/asterisk/users.conf': [Sep 16 14:23:28] == Found
[Sep 16 14:23:28] == Using SIP CoS mark 4
[Sep 16 14:23:28] == Parsing '/etc/asterisk/sip_notify.conf': [Sep 16 14:23:28] == Found
[Sep 16 14:23:28] WARNING[2170]: chan_sip.c:21487 handle_response_register: Got 404 Not found on SIP register to service 42530300@192.168.1.6, giving up
vicibox*CLI>

[geoff3dmg]

Just because asterisk says the peer is contactable doesn't mean it'll work. You need to attempt to dial a number to truely find out.

Code: Select all

[Sep 16 14:23:28] WARNING[2170]: chan_sip.c:21487 handle_response_register: Got 404 Not found on SIP register to service 42530300@192.168.1.6, giving up

Your VoIP provider said the registration details you used were not found. This usually indicates that they are incorrect.

[abdulnawas]

console dial 0505474840
[Sep 25 23:06:39] WARNING[21916]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Sep 25 23:06:39] NOTICE[21916]: console_video.c:137 console_video_start: voice only, console video support not present
[Sep 25 23:06:39] -- Executing [0505474840@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 25 23:06:39] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 25 23:06:39] -- Executing [0505474840@default:2] Dial("Console/dsp", "/0505474840") in new stack
[Sep 25 23:06:39] WARNING[21967]: channel.c:5754 ast_request: No channel type registered for ''
[Sep 25 23:06:39] WARNING[21967]: app_dial.c:2345 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
[Sep 25 23:06:39] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 25 23:06:39] -- Executing [0505474840@default:3] Hangup("Console/dsp", "") in new stack
[Sep 25 23:06:39] == Spawn extension (default, 0505474840, 3) exited non-zero on 'Console/dsp'
[Sep 25 23:06:39] -- Executing [h@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Sep 25 23:06:39] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 25 23:06:39] << Hangup on console >>

[geoff3dmg]

Code: Select all

[Sep 25 23:06:39] WARNING[21967]: channel.c:5754 ast_request: No channel type registered for ''
[Sep 25 23:06:39] WARNING[21967]: app_dial.c:2345 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)

The parameters you are passing to the Dial application in your dialplan are wrong.

[DomeDan]

post you "Globals String" from the carrier settings page

[abdulnawas]

[Sep 27 10:24:41] NOTICE[3168]: chan_sip.c:13875 sip_reg_timeout: -- Registration for '042530300.etisalat@192.168.1.6' timed out, trying again (Attempt #105)
[Sep 27 10:24:41] -- Got SIP response 500 "Server Internal Error" back from 192.168.1.6:5060

[abdulnawas]

vicibox*CLI> console dial 0505474840
[Sep 27 10:33:36] WARNING[9910]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Sep 27 10:33:36] NOTICE[9910]: console_video.c:137 console_video_start: voice only, console video support not present
[Sep 27 10:33:36] -- Executing [0505474840@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 27 10:33:36] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 27 10:33:36] -- Executing [0505474840@default:2] Dial("Console/dsp", "SIP/42530300/0505474840") in new stack
[Sep 27 10:33:36] == Using SIP RTP CoS mark 5
[Sep 27 10:33:36] -- Called SIP/42530300/0505474840
[Sep 27 10:33:36] -- Got SIP response 480 "No Routes Found" back from 192.168.1.6:5060
[Sep 27 10:33:36] -- SIP/42530300-00000000 is circuit-busy
[Sep 27 10:33:36] == Everyone is busy/congested at this time (1:0/1/0)
[Sep 27 10:33:36] -- Executing [0505474840@default:3] Hangup("Console/dsp", "") in new stack
[Sep 27 10:33:36] == Spawn extension (default, 0505474840, 3) exited non-zero on 'Console/dsp'
[Sep 27 10:33:36] -- Executing [h@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----19-----CONGESTION----------") in new stack
[Sep 27 10:33:36] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 27 10:33:36] << Hangup on console >>
[Sep 27 10:33:41] NOTICE[3168]: chan_sip.c:13875 sip_reg_timeout: -- Registration for '042530300.etisalat@192.168.1.6' timed out, trying again (Attempt #132)
[Sep 27 10:33:41] -- Got SIP response 500 "Server Internal Error" back from 192.168.1.6:5060
vicibox*CLI>

[abdulnawas]

vicibox*CLI> console dial 0505474840
[Sep 28 15:16:52] WARNING[18684]: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Sep 28 15:16:52] NOTICE[18684]: console_video.c:137 console_video_start: voice only, console video support not present
[Sep 28 15:16:52] -- Executing [0505474840@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 28 15:16:52] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 28 15:16:52] -- Executing [0505474840@default:2] Dial("Console/dsp", "SIP/42530300/0505474840") in new stack
[Sep 28 15:16:52] WARNING[18697]: chan_sip.c:5825 create_addr: Purely numeric hostname (42530300), and not a peer--rejecting!
[Sep 28 15:16:52] WARNING[18697]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Sep 28 15:16:52] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 28 15:16:52] -- Executing [0505474840@default:3] Hangup("Console/dsp", "") in new stack
[Sep 28 15:16:52] == Spawn extension (default, 0505474840, 3) exited non-zero on 'Console/dsp'
[Sep 28 15:16:52] -- Executing [h@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Sep 28 15:16:52] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 28 15:16:52] << Hangup on console >>

[abdulnawas]

Globals String: SIPDXB = SIP/42530300

[DomeDan]

do you really got your provider on the local network? 192.168.1.6?