i have a FreePBX 2.9.0.7 so i need to add trunk. but i have a only ip and port from my sip provider. so how can i add in PEER Details and Register String without username and password.
becouse its need ip, port, username, and password,
You don’t need to register if your provider has your IP address (and it is static) to send calls to. They must be validating on your IP.
You only need 1 entry for inbound and outbound:
context=from-trunk
type=friend
host=x.x.x.x (IP of your provider)
insecure=port,invite
disallow=all
allow=ulaw (or whatever CODEC your provider wants)
Skyking,
6 lines and 2 errors 
How can I convince you to stop suggesting to people type=friend ?
In addition insecure should not be used when there is no username/password authentication.
The correct syntax is
context=from-trunk (IP of your provider)
type=peer
host=x.x.x.x
disallow=all (or whatever CODEC your provider wants)
allow=ulaw
No, you have to have the insecure line to disable all auth checking.
You keep telling me not to use friend, I keep telling you it works 100% on two way trunks.
From the Digium sip.conf,sample:
[quote]Tip 2: Use separate inbound and outbound sections for SIP providers
664 ; (instead of type=friend) if you have calls in both directions
[/quote]
This is only a suggestion.
Dear Skyking,
I wish you could spent a few minutes reading users’ questions before blasting your boilerplate responses to almost every thread. Having said that I will address the concerns you are bringing up.
you have to have the insecure line to disable all auth checking
this is only true in case where there is auth checking. In the case being discussed here there is no auth checking as there is no password specified and only the IP address is checked in order to find a matching peer.
You keep telling me not to use friend, I keep telling you it works 100% on two way trunks.
So does type=peer, but it does not create user. You are simply over-specifying the trunk definition. By the same token you could always forward all ports to your box on the firewall and claim it works 100% of the time. Yes, it does, but is it secure ?
I am not sure why are you bringing the quote from sip.conf.sample, it clearly states you should not use type=friend.
Almost every thread? I participate in less than 15% of discussions.
I do stick with what works, and I understand the user creation. I can tell you that XO trunks are not requesting digest or any other type of authentication and they do require the insecure=port,invite. As far as the friend, I may need to revisit my use of this peer type. I need to do some controlled testing.
dear,
SkykingOH and obelisk
thanks for your reply i chak both config but it show me
CentOS-60-32-minimal*CLI> sip show peer bhaoo load
- Name : bhaoo
Secret :
MD5Secret :
Remote Secret:
Context : from-trunk
Subscr.Cont. :
Language : en
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 108.59.10.198
Addr->IP : 108.59.10.198:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : (none)
Codecs : 0x105 (g723|ulaw|g729)
Codec Order : (g729:20,g723:30,ulaw:20)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Do you have another question?
yes dear
i add trunk same as you say so its show me this message
Got SIP response 503 “Service Unavailable” back from 108.59.10.198:5060
[size=20][color=red]Don’t call me Dear[/size][/color]
WRT the 503 error, you have g.729 CODEC set as primary. Do you have g.729 licenses installed?
ok sorry for that <<<Don’t call me Dear>>>
now i still not installed licenses becouse still on larning stage,
but can we must need it on that expariment parpuse
g.729 requires licenses from Digium. That is why the calls are failing.
You need to remove the g.729 from your trunk config and from the SIP settings module.
thanks for helping
its working now, i install g729 code from digium so sound is very clear now
and also i add in trunk frefix becouse my voip provider give me ip port and frefix.
thanks for help again
do you need a license for g722 and g723. I am receiving this same error on a handful of calls throughout the day, so I disabled the g729 codec. Should this resolve the issue?
Log:
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:12] GosubIf(“SIP/102-000002cc”, “1?sub-flp-2,s,1”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] ExecIf(“SIP/102-000002cc”, “0?Set(TARGET_FLP_2=117737261469)”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:2] GotoIf(“SIP/102-000002cc”, “0?match”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:3] Return(“SIP/102-000002cc”, “”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:13] Set(“SIP/102-000002cc”, “OUTNUM=17737261469”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:14] Set(“SIP/102-000002cc”, “custom=SIP/voipms”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:15] ExecIf(“SIP/102-000002cc”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:16] ExecIf(“SIP/102-000002cc”, “0?Set(DIAL_TRUNK_OPTIONS=M(confirm))”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:17] Macro(“SIP/102-000002cc”, “dialout-trunk-predial-hook,”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] MacroExit(“SIP/102-000002cc”, “”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:18] GotoIf(“SIP/102-000002cc”, “0?bypass,1”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:19] GotoIf(“SIP/102-000002cc”, “0?customtrunk”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] pbx.c: – Executing [[email protected]:20] Dial(“SIP/102-000002cc”, “SIP/voipms/17737261469,300,”) in new stack
[Nov 10 13:56:20] VERBOSE[32152] netsock.c: == Using SIP RTP TOS bits 184
[Nov 10 13:56:20] VERBOSE[32152] netsock.c: == Using SIP RTP CoS mark 5
[Nov 10 13:56:20] VERBOSE[32152] app_dial.c: – Called voipms/17737261469
[Nov 10 13:56:22] VERBOSE[3035] chan_sip.c: – Got SIP response 503 “Service Unavailable” back from 64.120.22.242
[Nov 10 13:56:22] VERBOSE[32152] app_dial.c: – SIP/voipms-000002cd is circuit-busy
[Nov 10 13:56:22] VERBOSE[32152] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:21] NoOp(“SIP/102-000002cc”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:22] Goto(“SIP/102-000002cc”, “s-CONGESTION,1”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Goto (macro-dialout-trunk,s-CONGESTION,1)
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] Set(“SIP/102-000002cc”, “RC=34”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:2] Goto(“SIP/102-000002cc”, “34,1”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Goto (macro-dialout-trunk,34,1)
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] Goto(“SIP/102-000002cc”, “continue,1”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Goto (macro-dialout-trunk,continue,1)
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/102-000002cc”, “1?noreport”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Goto (macro-dialout-trunk,continue,3)
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:3] NoOp(“SIP/102-000002cc”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:4] Set(“SIP/102-000002cc”, “CALLERID(number)=102”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:5] Macro(“SIP/102-000002cc”, “outisbusy,”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] Progress(“SIP/102-000002cc”, “”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:2] GotoIf(“SIP/102-000002cc”, “0?emergency,1”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/102-000002cc”, “0?intracompany,1”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] pbx.c: – Executing [[email protected]:4] Playback(“SIP/102-000002cc”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
[Nov 10 13:56:22] VERBOSE[32152] file.c: – <SIP/102-000002cc> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
[Nov 10 13:56:24] VERBOSE[32152] file.c: – <SIP/102-000002cc> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
[Nov 10 13:56:26] VERBOSE[32152] pbx.c: – Executing [[email protected]:5] Congestion(“SIP/102-000002cc”, “20”) in new stack
[Nov 10 13:56:26] WARNING[32152] channel.c: Prodding channel ‘SIP/102-000002cc’ failed
[Nov 10 13:56:26] VERBOSE[32152] app_macro.c: == Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/102-000002cc’ in macro ‘outisbusy’
[Nov 10 13:56:26] VERBOSE[32152] pbx.c: == Spawn extension (from-internal, 17737261469, 5) exited non-zero on ‘SIP/102-000002cc’
[Nov 10 13:56:26] VERBOSE[32152] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/102-000002cc”, “”) in new stack
[Nov 10 13:56:26] VERBOSE[32152] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/102-000002cc’
It did not… any ideas? Another issue I am experiencing is the call connecting but I hear no ringtone, the person I’m calling just starts saying “hello? hello?”