@agasasterisk wrote:

Hi All,

I am trying to create an Inbound route destined to a Ring Group through a SIP trunk. I am able to call the extensions, but unable to call a Ring Group or an IVR through the Inbound Route. I am really not sure, what i am missing. When the DID for the IVR or Ring Group is called, getting the message from the Asterisk that "the call cannot be completed, please check your number".

The Inbound Route configuration for the IVR :-

1. DID Number : 2000
2. Ring Groups : RG<600>

SIP Peer details :-

host=20.1.1.170
type=friend
port=5060
nat=no
disallow=all
allow=ulaw,alaw
qualify=yes
canreinvite=yes
context=from-trunk

When 2000, is dialled, the DID in the SIP Invite is the same, but still getting the same error message.

SIP Logs :-

Invite to the DID 2000 for Ring Group ---->
100 Trying <-----
183 Session Progess <----- (Playing the error message)

---

<--- SIP read from UDP:20.1.1.170:5060 --->
INVITE sip:2000@20.1.1.58:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c
From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To:
Date: Fri, 11 Sep 2015 14:06:41 GMT
Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1376285696-0000065536-0000002594-2852192532
Session-Expires: 1800
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: ;bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 198

v=0
o=CiscoSystemsCCM-SIP 787014 1 IN IP4 20.1.1.170
s=SIP Call
c=IN IP4 20.1.1.170
t=0 0
m=audio 25986 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (22 headers 9 lines) ---
Sending to 20.1.1.170:5060 (no NAT)
Sending to 20.1.1.170:5060 (no NAT)
Using INVITE request as basis request - 52087400-5f21dff1-354b2-aa010114@20.1.1.170
Found peer '2723' for '2723' from 20.1.1.170:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 20.1.1.170:25986
Looking for 2000 in from-internal (domain 20.1.1.58)
list_route: hop:

<--- Transmitting (NAT) to 20.1.1.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060
From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To:
Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170
CSeq: 101 INVITE
Server: FPBX-12.0.76(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Length: 0

<------------>
Audio is at 16598
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 20.1.1.170:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060
From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To: ;tag=as3e6a1653
Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170
CSeq: 101 INVITE
Server: FPBX-12.0.76(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Require: timer
Content-Length: 228

v=0
o=root 881046367 881046367 IN IP4 20.1.1.58
s=Asterisk PBX 11.19.0
c=IN IP4 20.1.1.58
t=0 0
m=audio 16598 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
=====================================================

Anything i am missing here ?

Thanks a lot !

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@avayax wrote:

We are thinking of getting a few Sipstation trunks.
We can source-restrict SIP 5060 to a SIPstation IP address or domain name, so that's good.
We don't feel comfortable opening RTP ports to the internet at large though . Although there is no threat of hacking into our system, Dos attacks would still be possible.

What is your take/experience on this, and how does Asterisk deal with a ton of unauthorized RTP traffic?

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@Richi_Nbg wrote:

Hello,
Is it possible to directly call someone with the Ext. included?

So for example my number would be 1 800 5000. And My ext is 212.
So is it possible that someone from outside calls 1 800 5000 212 to get directly to me?
Or do I need to setup a IVR with "Call Direct Ext"?

I believe it is not possible since I only own numbers from 1 800 5000 0 - 1 800 5000 10 to do that but I would be happy if someone could clarify that. Thanks!

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@shillamus wrote:

SIP Registering. Rings External Phone. No Audio/Sound

FreePBX 12.0.76.2
Asterisk 11.19.0
Linux 2.6.32-431.el6.i686

I am using SIP Station trunks. The sips are registering.
I am able to ring my cell phone from the handset But there is no audio from the handset

The SIP trunks are in use on a second server and working fine. This PBX Server is running version FreePBX 2.9.0.14
I copied the trunk data and have changed a few things in my updated server

I have not proceeded to incoming routes yet.

I have configured a Flowroute trunk and have it working on the updated server fine

Peer details for the non working trunks are:

context=from-trunk
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=xxxxxx
secret=yyyyyy
host=trunk1.freepbx.com
disallow=all
allow=ulaw

some errors and warnings are in the log file
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context 'from-internal-xfer' tries to include nonexistent context 'from-internal-custom'
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context 'from-internal-noxfer' tries to include nonexistent context 'from-internal-noxfer-custom'
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context 'from-pstn' tries to include nonexistent context 'from-pstn-custom'
[2015-10-06 10:37:36] WARNING[9304] pbx.c: Context 'from-internal-additional' tries to include nonexistent context 'ext-meetme'

[2015-10-06 10:37:36] ERROR[9304] res_clialiases.c: res_clialiases configuration file 'cli_aliases.conf' not found

[2015-10-06 10:37:36] ERROR[9304] res_config_ldap.c: Cannot load configuration file: res_ldap.conf
[2015-10-06 10:37:36] NOTICE[9304] res_config_ldap.c: Cannot reload LDAP RealTime driver.

[2015-10-06 10:37:36] ERROR[9304] res_config_sqlite3.c: Missing config file 'res_config_sqlite3.conf'

[2015-10-06 10:37:36] ERROR[9304] phone_message.c: Unable to build dialplan routing - invalid license
[2015-10-06 10:37:36] WARNING[9304] res_digium_phone.c: No Valid DPMA License found. Module is loaded but disabled. Please reload module once valid license is installed.

Thanks of anyone can help with this issue. I have been unable to find many threads related to the issue.

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@avayax wrote:

Is it true that "unlimited" SIPSTATION trunks are actually not unlimited, but you can only make 3000 combined inbound and outbound calls in a month on one trunk?
We are making around 30000 minutes of outbound toll calls a month, but don't need 10 trunks.

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@eric wrote:

I saw an older post about this issue but didn't see a solution. Is there a module I need to purchase so that our phones display name and number of inbound call? Also, is there a software out there that will connect the phone system to our contact database so that it will query and show the names on the CID?

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@AaronGustafson wrote:

I have 4 trunks from SIPStation on my FreePBX install and the fourth trunk is not working. I even went so far as to do a complete reinstall (Needed to clean it up anyways). When I do a capacity test, I can initiate three simultaneous calls but when I go to do the fourth it says all circuits are busy... Can anyone point me in the right direction?

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@gprimr1 wrote:

I've been using SIPStation for a couple months now with a lot of success. Lately some members (we are a fire house) have reported getting 3 rings then a fast busy when making calls.

I did some research and it says that would indicate that means it couldn't find a path out. I'm thinking maybe it was an incoming call at the same time, but I have the option enabled to allow per minute overages on the trunk (we have 1 trunk)

Anything I could be missing?

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@andmore wrote:

We just switched our VOIP line to SIPStation and ever since we made the switch we are no longer getting the CID (CNAME) information. With my old provider I could see the information with no issue but since the switch to SIPStation we can not. We have configured the SIPStation trunk using the SIPSTATION module and have a very simple configuration.

We are currently running FreePBX 13.0.51 / Asterisk Version: 13.5.0.

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@TX_RX wrote:

I just added an additional trunk to my Sip Station Account, when I added the third one I expected to see the dashboard update to show the new trunk, however it did not.

I have confirmed that the trunk is there and I can use it, we did a test to three outside lines at once.

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@Dunmarie wrote:

I have just download the latest version of FreePBX Distro. Version FreePBX 13.0.70
Installation went without any issues.
However when trying to setup a new Trunk, I get the following error:

Reload failed because retrieve_conf encountered an error: 1
exit: 1
Whoops\Exception\ErrorException: Invalid argument supplied for foreach() in file /var/www/html/admin/modules/core/functions.inc.php on line 4216
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/core/functions.inc.php:4216
2. Whoops\Run->handleError() /var/www/html/admin/modules/core/functions.inc.php:4216
3. core_devices_get_user_mappings() /var/www/html/admin/modules/core/functions.inc.php:42
4. core_conf->map_dev_user() /var/www/html/admin/modules/core/functions.inc.php:452
5. core_conf->generate_sip_additional() /var/www/html/admin/modules/core/functions.inc.php:103
6. core_conf->generateConf() /var/www/html/admin/libraries/BMO/FileHooks.class.php:65
7. FreePBX\FileHooks->processOldHooks() /var/www/html/admin/libraries/BMO/FileHooks.class.php:24
8. FreePBX\FileHooks->processFileHooks() /var/lib/asterisk/bin/retrieve_conf:831

1 error(s) occurred, you should view the notification log on the dashboard or main screen to
check for more details.

When going to the Dashboard it doesn't give any extra info:

retrieve_conf failed, config not applied
Reload failed because retrieve_conf encountered an error: 1
1 minute, 28 seconds, ago

fwconsole ma list
No repos specified, using: [standard,commercial] from last GUI settings

+---------------------+--------------+-----------------------------------+
| Module | Version | Status |
+---------------------+--------------+-----------------------------------+
| accountcodepreserve | 13.0.2 | Enabled |
| announcement | 13.0.4 | Enabled |
| arimanager | | Not Installed (Locally available) |
| asterisk-cli | 13.0.3 | Enabled |
| asteriskinfo | 13.0.6 | Enabled |
| backup | 13.0.21.5 | Enabled |
| blacklist | 13.0.7 | Enabled |
| builtin | | Enabled |
| callback | 13.0.5 | Enabled |
| callforward | 13.0.4 | Enabled |
| callrecording | 13.0.9 | Enabled |
| callwaiting | 13.0.4 | Enabled |
| campon | 13.0.3 | Enabled |
| cdr | 13.0.23 | Enabled |
| cel | | Not Installed (Locally available) |
| certman | 13.0.12 | Enabled |
| cidlookup | | Not Installed (Locally available) |
| conferences | 13.0.17 | Enabled |
| conferencespro | 13.0.18 | Enabled |
| contactmanager | 13.0.17 | Enabled |
| core | 13.0.48 | Enabled |
| customappsreg | 13.0.4.4 | Enabled |
| dashboard | 13.0.19 | Enabled |
| daynight | 13.0.9 | Enabled |
| dictate | 13.0.4 | Enabled |
| directory | 13.0.10 | Enabled |
| disa | 13.0.5 | Enabled |
| donotdisturb | 13.0.3 | Enabled |
| endpoint | 13.0.32 | Enabled |
| extensionroutes | 13.0.6 | Enabled |
| featurecodeadmin | 13.0.5 | Enabled |
| findmefollow | 13.0.27 | Enabled |
| framework | 13.0.70 | Enabled |
| freepbx_ha | 13.0.7.2 | Enabled |
| fw_langpacks | 12.0.7 | Enabled |
| iaxsettings | 13.0.3 | Enabled |
| infoservices | 13.0.1 | Enabled |
| irc | 2.11.0.6 | Enabled |
| ivr | 13.0.17 | Enabled |
| languages | 13.0.5 | Enabled |
| logfiles | 13.0.7 | Enabled |
| manager | 13.0.2.5 | Enabled |
| miscapps | 13.0.2.2 | Enabled |
| miscdests | 13.0.2 | Enabled |
| music | 13.0.13 | Enabled |
| outroutemsg | 13.0.2 | Enabled |
| pbdirectory | 2.11.0.4 | Enabled |
| phonebook | 13.0.5.2 | Enabled |
| phpinfo | 13.0.2 | Enabled |
| pinsets | 13.0.5 | Enabled |
| presencestate | 13.0.4 | Enabled |
| printextensions | 13.0.3 | Enabled |
| queueprio | 13.0.2 | Enabled |
| queues | 13.0.19 | Enabled |
| recording_report | 13.0.17 | Enabled |
| recordings | 13.0.21 | Enabled |
| ringgroups | 13.0.14 | Enabled |
| setcid | 13.0.4 | Enabled |
| sipsettings | 13.0.17 | Enabled |
| sipstation | 13.0.13.7 | Enabled |
| sms | 13.0.5 | Enabled |
| soundlang | 13.0.8 | Enabled |
| speeddial | 2.11.0.3 | Enabled |
| sysadmin | 13.0.41 | Enabled |
| timeconditions | 13.0.15 | Enabled |
| tts | 13.0.6 | Enabled |
| ttsengines | 13.0.4.1 | Enabled |
| ucp | 13.0.22 | Enabled |
| userman | 13.0.50 | Enabled |
| vmblast | 13.0.7 | Enabled |
| voicemail | 13.0.33 | Enabled |
| weakpasswords | 13.0.1alpha1 | Enabled |
+---------------------+--------------+-----------------------------------+

Any ideas on how to fix this?

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@Rudy44 wrote:

Trying out the FreePBX for the first time. So far fairly successful. Internally things are running well. Signed up for the two week trial for SIPStation. No issues there (just took a while to read that it was automatically configured for me)

However now trying to make outside calls. "All circuits are busy" is all I get. The SIP trunk is not configured. So ok go to set it up. Missing information. Namely "Peer Details". Discovered that the SIPStation module should have a new display mode which lists these details and the tab is labeled "Local Settings". I do not have this tab. When I log into the SIPStation store, (through the module) I am able to view my Phone# and other account details. Just not the "Local Settings"

Tried reinstalling the module (12.0.8.8 ( FYI my FreePBX is 12.0.76.2)) This did not help. Please advise.

Thanks.

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@mvogel4949 wrote:

What are the costs of SMS using SIP Station on top of the price you are paying for the sip trunks? Unlimited SMS messages included? Thanks

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@hatrickwah wrote:

Hi all, I've been running FreePBX solid for about 2 years. Recently we've run into serious stability issues with our primary WAN connection (Charter), dropping out briefly for a minute or so and then return. As a result I've had a dedicated DSL connection brought in from a new provider. I'm running pFsense for my router, so no problem facilitating load balancing and such for my users. I also changed the routing of all traffic pointed to my PBX box as well as all the RTD and SIP traffic to go across the DSL. The SIP Settings detects the new IP no problem, however when I go into Sipstation module, it reflects both of my external addresses, my contact IP as the new IP, but my network IP as the old.
Is the Network info pinging data via a port or scanning what other devices are seeing? This has me scratching my head, as I said, all the traffic to the server is routed via DSL, and all the RTD and SIP traffic is as well.
Any insights would be greatly appreciated.

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@rudycalces wrote:

Hi,

I Have two box server PBX, I want to connected one side box to other box server with SIP trunk, not using SIP registered trunk
anyone can help ?

one box IP : 192.168.1.11
other box IP : 192.168.1.12

how to config ?

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@imcontreras wrote:

This is not a technical issue, I want to find out what providers offer better international call termination.

My current provider it's not very good, many times cannot connect to the destination or explicitly deny the connection because it's to expensive for them.

Who's using heavy international and what provider are you using? any recommendation to try?

Order:
Quality
Calling destinations
Price

My costumer it's in the entertainment business and call to many places must in south america, Europe (central and west), Russia and Caribe (including Cuba)

Thanks for any input

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@runtime wrote:

Hello everyone,

i'm new to freepbx, i'm trying to configure a Topex MobiLink IP - VoIP-GSM Gateway with freepbx. At this time i can correctly receive calls from the GSM box, but when i try to use GSM trunk we receive "service unavailable". Could it be in Peer details the problem? i use this parameters:

type=peer
host=
fromdomain=
trustpid=yes
qualify=yes
port=5060
disallow=all
allow=alaw&ulaw
directmedia=no
insecure=port,invite
context=from-pstn
dtmfmode=rfc,2833

Thank you all for advices.

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@xxoorr wrote:

Hello,
I am experiencing difficulties making proper configuration on the remote PBX phone.
We are using PBX system in our main office and with the PBX IP I have remote phone configured on port 5060. The problem is that a lot of 'dead calls' are under way - random 3 and 4 digit phone numbers are calling and no one is on the line. For that reason I bought SonicWall TZ300 which is connected into its WAN port and the connection is coming from an internal switch. The Grandstream phone is on the same switch. What rule should I create to allow only the PBX IP from the main office coming through the firewall and how to deny all other traffic so we won't get any more of these bot calls?

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@Jordack wrote:

I have a couple of the FaxStation CPE's and need to configure my firewall to let them through.

I think i have the ports 443/TCP and 4332/UDP correct

Just need IP ranges /FQDN. They don't look to use the same hostnames as sip trunks. There doesn't seem to be goof PTR records for the IP's its logging.

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@msinko wrote:

I got a call today and realized that I didn't have audio in either direction. When I checked my sipstation connectivity under Connectivity >> SIPSTATION I found that my "Contact IP" is showing my LAN address as opposed to my WAN address. My "Network IP" is showing the correct WAN address. Both are highlighted in yellow and right below it says:

"Warning: The SIP Contact header is not set to your WAN IP. It is set to your internal private IP behind NAT. The gateway will attempt to decipher your proper address but your configuration is incorrect. You should review the NAT settings in the Asterisk SIP Settings module, or sip_nat.conf if not using that module."

I checked my NAT settings in "Asterisk SIP Settings" and everything seems to be correct. Nothing has been changed. My external IP is correctly detected and my local network is properly defined as 192.168.0.0/24.

Everything was working fine a week ago when I did my last test after upgrading my firewall hardware. All required ports are open and freepbx was restarted after the change.

Question: How is the "Contact IP determined or set? And what do I change to make this match my external address?

I'm assuming this has something to do with my new router hardware but I'm not sure what I'm missing.

Thanks in advance for the help.

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