Hi I am new in FreePBX, I installed FreePBX 2.8.1.4

All extension cannot login to FreePBX, can anyone advise how to debug the problem, thanks in advance

Reading through the Terms of Service pdf it states the following:
"This service it to be registered to a single PBX/Phone System that belongs to a single company for their internal use only."

We are looking at implementing a system which will include two FreePBX boxes (1 at each location) connected via an IAX2 trunk. Does this fall into the "single PBX/Phone System" or will we need to purchase 2 SIP Trunks in this scenario?

Hi I am new in FreePBX, I installed FreePBX 2.8.1.4

All extension cannot login to FreePBX, can anyone advise how to debug the problem, thanks in advance

Reading through the Terms of Service pdf it states the following:
"This service it to be registered to a single PBX/Phone System that belongs to a single company for their internal use only."

We are looking at implementing a system which will include two FreePBX boxes (1 at each location) connected via an IAX2 trunk. Does this fall into the "single PBX/Phone System" or will we need to purchase 2 SIP Trunks in this scenario?

I have an asterisk setup and getting multiple trunks. I have one DID (main number), that routes to a queue ringing all agents. What happens to additional incoming calls when somebody calls that primary did while another agent is on the phone. Pretty much i want to make sure i can have like 4 concurrent calls to the same DID if i have 4 trunks. Does freepbx automatically free up the DID as soon as it routes to an extension or queue?

Sipstation is caching the IP from my old FreePBX installation
For whatever reason the IP Address in the database will not update to the new public IP of my current FreePBX

I have removed, using the sipstation plugin, the keys and trunks on the old PBX

On the new PBX I have added the key and then the trunks are generated but the IP Address still shows the old IP of the old server.

Please advise how to request that the cache be cleared.
Thanks
Mike

At the moment my configuration is FreeBpx and A2Billing. The trunk configuration I have got is surname, password, and address (http://222.000.00.00)...the problem is I have not got any idea how to set trunk (in FreePbx or A2Billing or in both) up with this trunk info they (trunk provider) provided me with the assurance that they added my ip as a trusted one with their server.

It would be a great help if you could tell me where (FreePbx or A2Billin or in both?) exactly I need to set up trunk (and how)

Can you please help.

Kind Regards

HI

My ISP service provide me a router and placed a Alcatel-Lucent phone (IP phone) to me. this Device is working as IP phone for incoming call. and it configuration is below

IP 192.168.255.195
sub 255.255.255.224
router 192.168.255.193
TFTP1 192.168.92.246
TFTP2 255.255.255.255

CPU1 192.168.92.246
CPU2 255.255.255.255

So please let me know how can i configure this ip in my freePbx to receive calls from FreePBX

hi there
PLEASE HELP MEE...................

How can I configure a SIP trunk between FreePBX and Exchange2013 for Unified messaging?
My Exchange IP Address: 10.10.150.59
My FreeBPX IP Address: 10.10.150.86

I created one before, with the following parameters:

host=10.10.150.59
transport=tcp
port=5065
insecure=very
type=friend
fromdomain=10.10.150.86
context=from-internal
qualify=yes
canreinvite=yes

but it doesn't work for me. I receiving the following Error when I am in Asterisk -r:

ERROR[12331]: tcptls.c:451 ast_tcptls_client_start: Unable to connect SIP socket to 10.10.150.59:5065: Connection timed out

ERROR[12499]: tcptls.c:451 ast_tcptls_client_start: Unable to connect SIP socket to 10.10.150.59:5065: Connection timed out

please please help me for this case.
thank you....

I´m trying to register this SIP Provider:
webcalldirect.com
voipbuster.com
...both not working!
Please help me to get thsi trunk registered -
-------------------------------------------------------
Peer Details:

type=peer
context=inbound
username=XXXXXXXXXXuserXXXXXXXXXXX
secret=XXXXXXsecretXXXXXXXXXXXXXX
fromuser=XXXXXXXXXXuserXXXXXXXXXXX
fromdomain=sip.webcalldirect.com
host=sip.webcalldirect.com
qualify=no
insecure=very
canreinvite=no
allow=all

---------------------------------------------------

---
[2013-11-01 23:32:02] NOTICE[1793] chan_sip.c: -- Registration for 'jerry_theonlyone_fl@webcalldirect.com' timed out, trying again (Attempt #22)
[2013-11-01 23:32:02] VERBOSE[1793] chan_sip.c: Really destroying SIP dialog '4172dad0268961331d788cc74b0ba258@[::1]' Method: REGISTER
[2013-11-01 23:32:02] VERBOSE[1793] chan_sip.c: Retransmitting #1 (NAT) to 77.72.169.9:5060:
REGISTER sip:webcalldirect.com SIP/2.0
Via: SIP/2.0/UDP 178.82.243.231:5060;branch=z9hG4bK7d73fb46;rport
Max-Forwards: 70
From: ;tag=as478b1c4f
To:
Call-ID: 4172dad0268961331d788cc74b0ba258@[::1]
CSeq: 123 REGISTER
User-Agent: FPBX-2.11.0(11.5.1)
Expires: 120
Contact:
Content-Length: 0

--------------------------------

---
[2013-11-01 23:32:02] NOTICE[1793] chan_sip.c: -- Registration for 'jerry_theonlyone_fl@webcalldirect.com' timed out, trying again (Attempt #22)
[2013-11-01 23:32:02] VERBOSE[1793] chan_sip.c: Really destroying SIP dialog '4172dad0268961331d788cc74b0ba258@[::1]' Method: REGISTER
[2013-11-01 23:32:02] VERBOSE[1793] chan_sip.c: Retransmitting #1 (NAT) to 77.72.169.9:5060:
REGISTER sip:webcalldirect.com SIP/2.0
Via: SIP/2.0/UDP 178.82.243.231:5060;branch=z9hG4bK7d73fb46;rport
Max-Forwards: 70
From: ;tag=as478b1c4f
To:
Call-ID: 4172dad0268961331d788cc74b0ba258@[::1]
CSeq: 123 REGISTER
User-Agent: FPBX-2.11.0(11.5.1)
Expires: 120
Contact:
Content-Length: 0

I have three servers AsteriskNOW, currently I've only managed to connect two of them.
Server A, connected by a sip trunk to Server B.
Server B, connected to the server another sip trunk C.
A server telephones, telephones communicate with server B.
B server telephones, telephones communicate with server C.
I need the phone from Server A phone to communicate with server C. that should the server configuration B, to transfer calls from Server A to Server C?

Hi

I have realy hard tryed to setup a trunk to a Swedish PSTN called Tele2.
They don´t like registering of the trunk
Authentication should be:
HTTP MD5 authentication

Is there someone who has a solution to this?

Hello,
Sorry for my bad english I am from latin-america... well the ponit is that Ihave already 2 distro free pbx up,Icreadted extension I can do and get calls,the problem is when I try to create trunks between 2 PBX.I search and found inf here also a I did everything i read. but there is a mesage "tehservic is busy now please try to call later" something like that,, mi server´s ip address are
- server one : 192.168.1.253... (extensions 6001 to 6009)
- server two : 192.168.2.253.... (extension 3001 to 3009)

please help me,I thing the wrong is the outbound route in the part of dial patterns.
thanks.

I initially reviewed this thread (http://www.freepbx.org/forum/freepbx/general-help/new-to-free-pbx-cannot-make-outgoing-calls-via-trunk#comment-40447) and I’m having the same issue, cannot make outbound calls. I can receive incoming calls from the outside just fine and I can place and receive internal calls to all my extensions.

I’m currently running FreePBX 4.211.64-9, Asterisk 11.6, and Centos 6.4

I have enabled EndPoint Manager and I have a SIPSTATION trunk with one DID. I do have the UDP ports (5060, 10000-20000) forwarded to the pbx server. In SIPSTATION, “Run Firewall Test” I have all green lights and status PASS, so I don’t believe there is a firewall issue?

My network connection is as follows:
Server & Cisco 7940/7960 phones Cisco Switch (SFE2000p)  D-link DIR-655. I do have a static IP from my ISP and my Switch has “fast port” enabled on the ports the phones connect to. My Cisco phones are running Sip 8.12 and endpoint manager configured them perfectly (side note: I would agree with SkykingOH on taking CCM outback and shooting it, I have read too many posts).

Below is a copy of the Asterisk CLI output. Please let me know your thoughts and ideas as to what the issue may be or if you need any additional info. I appreciate all the posts by many of the “regulars” as you have helped me in my setup process.

>[root@localhost ~]# asterisk -RvvvT
>[Dec 10 19:00:44] Asterisk 11.6.0, Copyright (C) 1999 - 2013 Digium, Inc. and ot hers.
>[Dec 10 19:00:44] Created by Mark Spencer
>[Dec 10 19:00:44] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
>[Dec 10 19:00:44] This is free software, with components licensed under the GNU General Public
>[Dec 10 19:00:44] License version 2 and other licenses; you are welcome to redistribute it under
>[Dec 10 19:00:44] certain conditions. Type 'core show license' for details.
>[Dec 10 19:00:44] ============================================================== ===========
>[Dec 10 19:00:44] Connected to Asterisk 11.6.0 currently running on localhost (pid = 2040)
>== Using SIP RTP TOS bits 184
>== Using SIP RTP CoS mark 5
> -- Executing [8013905200@from-internal:1] Macro("SIP/511-00000143", "user-callerid,LIMIT") in new stack
> -- Executing [s@macro-user-callerid:1] Set("SIP/511-00000143", "TOUCH_MONITOR=1386727268.323") in new stack
> -- Executing [s@macro-user-callerid:2] Set("SIP/511-00000143", "AMPUSER=511") in new stack
> -- Executing [s@macro-user-callerid:3] GotoIf("SIP/511-00000143", "0?report") in new stack
> -- Executing [s@macro-user-callerid:4] ExecIf("SIP/511-00000143", "1?Set(REALCALLERIDNUM=511)") in new stack
> -- Executing [s@macro-user-callerid:5] Set("SIP/511-00000143", "AMPUSER=511") in new stack
> -- Executing [s@macro-user-callerid:6] Set("SIP/511-00000143", "AMPUSERCIDNAME=Office") in new stack
> -- Executing [s@macro-user-callerid:7] GotoIf("SIP/511-00000143", "0?report") in new stack
> -- Executing [s@macro-user-callerid:8] Set("SIP/511-00000143", "AMPUSERCID=511") in new stack
> -- Executing [s@macro-user-callerid:9] Set("SIP/511-00000143", "__DIAL_OPTIONS=Ttr") in new stack
> -- Executing [s@macro-user-callerid:10] Set("SIP/511-00000143", "CALLERID(all)="Office"<511>") in new stack
> -- Executing [s@macro-user-callerid:11] GotoIf("SIP/511-00000143", "0?limit") in new stack
> -- Executing [s@macro-user-callerid:12] ExecIf("SIP/511-00000143", "1?Set(GROUP(concurrency_limit)=511)") in new stack
> -- Executing [s@macro-user-callerid:13] ExecIf("SIP/511-00000143", "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s@macro-user-callerid:14] GosubIf("SIP/511-00000143", "7?sub-ccss,s,1(from-internal,8013905200)") in new stack
> -- Executing [s@sub-ccss:1] ExecIf("SIP/511-00000143", "0?Return()") in new stack
> -- Executing [s@sub-ccss:2] Set("SIP/511-00000143", "CCSS_SETUP=TRUE") in new stack
> -- Executing [s@sub-ccss:3] GosubIf("SIP/511-00000143", "0?monitor_config,1(from-internal,8013905200):monitor_default,1(from-internal,8013905200)") in new stack
> -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/511-00000143", "0?is_exten") in new stack
> -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/511-00000143", "") in new stack
> -- Executing [monitor_default@sub-ccss:3] Return("SIP/511-00000143", "FALSE") in new stack
> -- Executing [s@macro-user-callerid:15] GotoIf("SIP/511-00000143", "1?continue") in new stack
> -- Goto (macro-user-callerid,s,28)
> -- Executing [s@macro-user-callerid:28] Set("SIP/511-00000143", "CALLERID(number)=511") in new stack
> -- Executing [s@macro-user-callerid:29] Set("SIP/511-00000143", "CALLERID(name)=Office") in new stack
> -- Executing [s@macro-user-callerid:30] Set("SIP/511-00000143", "CDR(cnum)=511") in new stack
> -- Executing [s@macro-user-callerid:31] Set("SIP/511-00000143", "CDR(cnam)=Office") in new stack
> -- Executing [s@macro-user-callerid:32] Set("SIP/511-00000143", "CHANNEL(language)=en") in new stack
> -- Executing [8013905200@from-internal:2] Set("SIP/511-00000143", "ROUTEUSER=511") in new stack
> -- Executing [8013905200@from-internal:3] GotoIf("SIP/511-00000143", "1?outbound-4-5-6,8013905200,2:outbound-allroutes,8013905200,2") in new stack
> -- Goto (outbound-4-5-6,8013905200,2)
> -- Executing [8013905200@outbound-4-5-6:2] Set("SIP/511-00000143", "MOHCLASS=default") in new stack
> -- Executing [8013905200@outbound-4-5-6:3] Set("SIP/511-00000143", "_NODEST=") in new stack
> -- Executing [8013905200@outbound-4-5-6:4] Gosub("SIP/511-00000143", "sub-record-check,s,1(out,8013905200,)") in new stack
> -- Executing [s@sub-record-check:1] Set("SIP/511-00000143", "REC_POLICY_MODE_SAVE=") in new stack
> -- Executing [s@sub-record-check:2] GotoIf("SIP/511-00000143", "1?check") in new stack
> -- Goto (sub-record-check,s,7)
> -- Executing [s@sub-record-check:7] Set("SIP/511-00000143", "__MON_FMT=wav") in new stack
> -- Executing [s@sub-record-check:8] GotoIf("SIP/511-00000143", "1?next") in new stack
> -- Goto (sub-record-check,s,11)
> -- Executing [s@sub-record-check:11] ExecIf("SIP/511-00000143", "0?Return()") in new stack
> -- Executing [s@sub-record-check:12] ExecIf("SIP/511-00000143", "0?Set(__REC_POLICY_MODE=)") in new stack
> -- Executing [s@sub-record-check:13] GotoIf("SIP/511-00000143", "0?out,1") in new stack
> -- Executing [s@sub-record-check:14] Set("SIP/511-00000143", "__REC_STATUS=INITIALIZED") in new stack
> -- Executing [s@sub-record-check:15] Set("SIP/511-00000143", "NOW=1386727268") in new stack
> -- Executing [s@sub-record-check:16] Set("SIP/511-00000143", "__DAY=10") in new stack
> -- Executing [s@sub-record-check:17] Set("SIP/511-00000143", "__MONTH=12") in new stack
> -- Executing [s@sub-record-check:18] Set("SIP/511-00000143", "__YEAR=2013") in new stack
> -- Executing [s@sub-record-check:19] Set("SIP/511-00000143", "__TIMESTR=2013 1210-190108") in new stack
> -- Executing [s@sub-record-check:20] Set("SIP/511-00000143", "__FROMEXTEN=511") in new stack
> -- Executing [s@sub-record-check:21] Set("SIP/511-00000143", "__CALLFILENAME=out-8013905200-511-20131210-190108-1386727268.323") in new stack
> -- Executing [s@sub-record-check:22] Goto("SIP/511-00000143", "out,1") in new stack
> -- Goto (sub-record-check,out,1)
> -- Executing [out@sub-record-check:1] ExecIf("SIP/511-00000143", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
> -- Executing [out@sub-record-check:2] GosubIf("SIP/511-00000143", "0?record,1(exten,8013905200,511)") in new stack
> -- Executing [out@sub-record-check:3] Return("SIP/511-00000143", "") in new stack
> -- Executing [8013905200@outbound-4-5-6:5] Macro("SIP/511-00000143", "dialout-trunk,1,8013905200,,off") in new stack
> -- Executing [s@macro-dialout-trunk:1] Set("SIP/511-00000143", "DIAL_TRUNK=1") in new stack
> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/511-00000143", "0?sub-pincheck,s,1()") in new stack
> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/511-00000143", "0?disable trunk,1") in new stack
> -- Executing [s@macro-dialout-trunk:4] Set("SIP/511-00000143", "DIAL_NUMBER=8013905200") in new stack
> -- Executing [s@macro-dialout-trunk:5] Set("SIP/511-00000143", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
> -- Executing [s@macro-dialout-trunk:6] Set("SIP/511-00000143", "OUTBOUND_GROUP=OUT_1") in new stack
> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/511-00000143", "1?nomax") in new stack
> -- Goto (macro-dialout-trunk,s,9)
> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/511-00000143", "0?skipoutcid") in new stack
> -- Executing [s@macro-dialout-trunk:10] Set("SIP/511-00000143", "DIAL_TRUNK_OPTIONS=Tt") in new stack
> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/511-00000143", "outbound-callerid,1") in new stack
> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/511-00000143", "0?Set(CALLERPRES()=)") in new stack
>-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/511-00000143", "0?Set(REALCALLERIDNUM=511)") in new stack
> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/511-00000143", "1?normcid") in new stack
> -- Goto (macro-outbound-callerid,s,6)
> -- Executing [s@macro-outbound-callerid:6] Set("SIP/511-00000143", "USEROUTCID=") in new stack
> -- Executing [s@macro-outbound-callerid:7] Set("SIP/511-00000143", "EMERGENCYCID=") in new stack
> -- Executing [s@macro-outbound-callerid:8] Set("SIP/511-00000143", "TRUNKOUTCID=8018953650") in new stack
> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/511-00000143", "1?trunkcid") in new stack
>-- Goto (macro-outbound-callerid,s,14)
>-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/511-00000143", "1?Set(CALLERID(all)=8018953650)") in new stack
> -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/511-00000143", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/511-00000143", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/511-00000143", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
> -- Executing [s@macro-outbound-callerid:18] Set("SIP/511-00000143", "CDR(outbound_cnum)=8018953650") in new stack
> -- Executing [s@macro-outbound-callerid:19] Set("SIP/511-00000143", "CDR(outbound_cnam)=") in new stack
> -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/511-00000143", "0?sub-flp-1,s,1()") in new stack
> -- Executing [s@macro-dialout-trunk:13] Set("SIP/511-00000143", "OUTNUM=8013905200") in new stack
> -- Executing [s@macro-dialout-trunk:14] Set("SIP/511-00000143", "custom=SIP/fpbx-1-IX6Xupk5Ar6X") in new stack
> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/511-00000143", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
> -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/511-00000143", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
> -- Executing [s@macro-dialout-trunk:17] Macro("SIP/511-00000143", "dialout-trunk-predial-hook,") in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/511-00000143", "") in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/511-00000143", "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/511-00000143", "1?Set(CONNECTEDLINE(num,i)=8013905200)") in new stack
> -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/511-00000143", "1?Set(CONNECTEDLINE(name,i)=CID:8018953650)") in new stack
> -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/511-00000143", "0?customtrunk") in new stack
> -- Executing [s@macro-dialout-trunk:22] Dial("SIP/511-00000143", "SIP/fpbx-1-IX6Xupk5Ar6X/8013905200,300,Tt") in new stack
>== Using SIP RTP TOS bits 184
>== Using SIP RTP CoS mark 5
>-- Called SIP/fpbx-1-IX6Xupk5Ar6X/8013905200
>[2013-12-10 19:01:08] WARNING[27477][C-000000d2]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/fpbx-1-IX6Xupk5Ar6X-00000144>to SIP/511-00000143
>== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/511-00000143' in macro 'dialout-trunk'
>== Spawn extension (outbound-4-5-6, 8013905200, 5) exited non-zero on 'SIP/511-00000143'

Can anybody please advise on the below scenario,

We are trying to link our Elastix PBX server with Avaya PBX ( S8800 PBX) through SIP Trunk. We used the following settings to communicate with Avaya.

ASTERISK
Create a SIP Trunk like this.
Trunk Name : IPO
Peer Details :
context=from-internal
host=AVAYA's IP
type=friend
qualify=yes
insecure=very
disallow=all
allow=alaw&ulaw&gsm

User Details :
type=user
host=AVAYA's IP
context=from-internal
disallow=all
allow=gsm

Create Outbound Route.
Route Name : IPOffice
Dial Patterns : 3XXX (According your AVAYA's extension format)
Trunk Squence : SIP/IPO
Under General Settings
Set "Allow Anonymous Inbound Sip Calls" to yes

Still not able to call, Shows "unreachable" in the Sip status.
And gives "circuit are busy rightnow" message while calling.

Please Assist !!

Thank you !!

I have two PBX boxes setup, at two locations that are connected via LAN. I've configured an IAX2 trunk connecting the two sites, and that is working correctly. I'm trying to setup DID's going through our main site A to site B.

I've setup an inbound route through site a to go to the trunk to site b and then setup an inbound route on site b to catch the DID and pass it through an extension but it just gives me an all circuits are busy when i dial the number.

Any suggestions?

My configuration is as follows:
running FreePBX 2.11 on both boxes, trunk is configured:

Peer Details:
host=x.x.x.x
username=pbx2
secret=****
type=peer
context=from-internal

User Details:
secret=*****
type=user
context=from-internal

and vice-versa on the other box.

Hello everyone, I have been working on this for several days and am stumped at this point...

My environment is:
- FreePBX installed on VM (VMWare ESXI Host)
- SipStation Trunks (used auto-config key code to setup)
- CentOS-based Firewall
- Dual Wan:
1) ppp0 - static with multiple IPs (low-bandwidth)
2) eth0 - dynamic cable connection (tons of bandwidth)
- traffic nat'ed via iptables (for vm in question)
- all 5060 traffic is routed through static wan interface, as well as all traffic from pbx internal ip (note: routed both just as a failsafe)
- "Asterisk Sip Settings" in freepbx gui set to NAT with static IP and internal network defined

I am trying to build an environment to host multiple FreePBX systems on VMWare ESXI box and have been having all kinds of issues, but at this point I think there may be something corrupt in freepbx or some networking nuance I am missing. I have set up several installs in similar environments and always did a 1-1 NAT to an available static IP from the internal IP. That has worked 100% of the time using the same exact firewall setup in all my other installs and it works in my current environment as well for one of my other FreePBX installs. The problem is that on this particular FreePBX install I am trying to avoid a 1-1 NAT setup so that later I can configure more than one system on the same ip and just nat different ports based on the host. What I am seeing now in FreePBX gui under Connectivity > SipStation, is the secondary trunk is registered but shows extern-ip of eth0 and contact ip of ppp0 (it should be ppp0) and the primary trunk shows contact ip and network ip of ppp0 connection like it should. On my firewall, eth0 is the "primary" interface but with nat rules set and if ONE trunk is registering via correct interface my firewall would not just consistently decide to send all requests for trunk2 registration out a different interface. That just doesn't make sense. I have rebooted and re-applied config several times just to make sure it took and had a chance to register again, but the secondary trunk still shows external-IP of eth0 under network IP. I am at a loss here... and any help would be greatly appreciated.

**On a side note, I was originally trying to get this to work using my dynamic connection and a dynamic host-name (I have one setup already) but switched to using the static connection to make it a bit cleaner for troubleshooting. My end goal is still to get it working on the dynamic cable connection. If anyone feels extra generous with their time or knows how to make that work instead of just getting it working on the ppp0 connection that would be awesome but seriously, any direction or help to get it working correctly on even the ppp0 connection would be much appreciated.

Thanks in advance!
-Cody

Good Day,

I have SIP trunks setup on my PBX however I would like my users to see the incoming caller ID all they see is "unknown" when the IVR passes a call through to them. I have been doing a lot of searching however I cant seem to locate any info that can help. Any assistance you guys can provide will be much appreciated.Currently running asterisk 1.8.24.0

I was wondering if there is something I can adjust for more line quality. I don't think it is the DSL connection, because if I use another provider directly from my phone without going through the Freepbx the quality is fine. Perhaps there are some pointers regarding this.

James

Hi All,

It's my first post in this forum

When I try to froward the sip trunk from my asterisk server to another one, I got this reply.

" Destination server is not asterisk they allow only g729codec,"

I have downloaded the g729codec and copied to modules directory:

==========================
c1*CLI> core show codecs
256 (1 << (0x100) audio g729 (G.729A)
========================

My extension.conf is pasting below:

=============================
exten => 123456,1,Verbose(******** Telecom FORWARDING *********)
exten => 123456,n,set(SIP_CODEC=g729)
exten => 123456,n,Dial(SIP/1.2.2.1/${EXTEN})
exten => h,1,Hangup()
============================

1.2.2.1 is my destination server, Call forward will work fine when they allow default codec.So I think its related to g729 codec, How can I check this,Please help me .