@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?
Posts: 1
Participants: 1
@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?
Posts: 2
Participants: 1
@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.
Posts: 4
Participants: 2
@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.
Posts: 3
Participants: 2
@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:8311 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, agofwconsole 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?
Posts: 3
Participants: 2
@rdesilva wrote:
Hi
I am new to FreePBX (8). We are currently running an older version of the application and I am trying to create a new server with FreePBX 13 and Asterisk 13.
When we try to connect one of our existing sip phones to the new server we see the following ICMP error:
udp port 5060 unreachableWhen I do a netstat -anp, I get the following:
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name
tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN 1293/mysqld
tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 2027/asterisk
tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN 971/sshd
tcp 0 36 10.10.1.43:22 10.10.5.142:62943 ESTABLISHED 22441/2
tcp 0 0 127.0.0.1:49678 127.0.0.1:5038 TIME_WAIT -
tcp6 0 0 :::80 :::* LISTEN 1396/apache2
tcp6 0 0 :::22 :::* LISTEN 971/sshd
udp 0 0 0.0.0.0:62921 0.0.0.0:* 755/dhclient
udp 0 0 0.0.0.0:44179 0.0.0.0:* 2027/asterisk
udp 0 0 0.0.0.0:68 0.0.0.0:* 755/dhclient
udp6 0 0 :::36608 :::* 755/dhclientI am using pjsip on this version and the port says it is supposed to bind to 5060 but as can be seen from the output above - this has not happened.
I also noticed on this version, that the full asterlisk log files are not being generated even though they are present in the logger_logfiles_additional.conf.
This is running ubuntu-14.04.01:
Linux iit-sipserver-02 4.2.0-34-generic #39~14.04.1-Ubuntu SMP Fri Mar 11 11:38:02 UTC 2016 x86_64 x86_64 x86_64 GNU/LinuxNeed some help to understand what could be going wrong which is difficult without the logs files.
Thanks
Ranil
Posts: 1
Participants: 1
@freak wrote:
How do I control the outbound caller id? When I place a call it shows "unavailable". I tried on the trunk and route but nothing seems to change it.
Posts: 1
Participants: 1
@pramirez wrote:
So far I have set up my IP phone with a static IP address and I have created an extension. I can login to my ip phone, freePBX local server, and my SIPstation account. When I call my DID number it doesn't ring and goes immediately to the prompt, "The person at extension XXXX is unavailable please leave a message after the tone."
I am using a T46G IP Phone as well as SIPstation.
I have tried to use the documentation but because I am new I am uncertain with what I am doing wrong. Please provide me with a troubleshooting strategy. I would really love to dig into the IVR after this.
Thank you all so much,
Phil
Posts: 7
Participants: 3
@thecooldude wrote:
I just switched our ISDN PRI line from Avaya G430 to Asterisk with Digium T133 and getting the following error.
[2015-01-25 12:22:29] WARNING[17313][C-000004c6]: channel.c:4860 ast_prod: Prodding channel 'SIP/192.168.3.9-00000511' failed
[2015-01-25 12:23:54] WARNING[17341][C-000004cb]: channel.c:4860 ast_prod: Prodding channel 'SIP/192.168.3.9-00000516' failedPlease advise as I'm getting this error frequently.
Posts: 2
Participants: 1
@Mathabathe wrote:
I have changed the outbound routes to use the Telkom (POTS)(DAHDI/g0) on freepbx and not g1 (voip)(DAHDI/g1) however when I dial it is still using voip .
Posts: 3
Participants: 3
@chattel wrote:
New admin here. I have deployed a Freepbx system using 1 trunk from Sipstation. We can make outbound calls as well as receive inbound calls. Works as expected.
My boss asked me yesterday what message would a inbound caller get if we were offline? Is it the "The number you have dialed is not in service, please check the number and try again." message or something else.
Sorry if this is a dumb q. I tried looking for a faq of different scenarios, but could not find it. If somebody has a link that explains the different scenarios, that would be great.
thx
Posts: 3
Participants: 2
@bobby_inline wrote:
New installation of an Elastix phone system and randomly, at least once a day - and on different extensions - incoming callers are hearing a ringing on their end. The ringing is loud and our staff cannot hear the user on the other the end. The incoming caller hangs up and calls back and the conversation is fine.
Phones are 10.250.20.X
Data is on 10.25.40.Xelastix 2.4.0 ( I was looking at Linux version - Unfortunately thats not the part I need help with)
Posts: 3
Participants: 2
@GSnover wrote:
A friend is using SIPStation trunks behind a Motorola Surfboard (SB6850) cable modem - if we enable the SIP ALG inbound works perfectly, but outbound fails with an authentication failure - if we disable the SIP ALG outbound completes the call (no auth error) but there is no audio.
Have you all seen this, and is there any way to work around it? I have tried lots of combinations with SIP NAT settings and ALG settings to no avail.
If not, I am thinking we are going to have to replace the modem with a dumb one that works right.
Thanks!
Posts: 4
Participants: 3
@sonny wrote:
I have two Freepbx 12 setup. Primary PBX is setup with SIPStation and I have setup warm spare PBX in different state. My warm spare backs up and restores daily, disables trunks and excludes NAT settings since it's running on a different IP address. I know I can go online to SIPStation site and put in the IP address for failover PBX (warm spare), but can anyone tell what other changes I would have to make on my warm spare PBX so my trunks/sipstation would work properly when primary PBX is down. We want my warm spare to take over the trunks automatically if our broadband or our primary PBX goes down. We have remote office users using Yealink phones and on SIP Server 1 we have entered Primary PBX IP adddress and on SIP Server 2 we entered Warm Spare PBX IP address. Any input or help in this matter will be greatly appreciated.
SM
Posts: 4
Participants: 3
@uncleaelfrich wrote:
Looking at my telephone bill and current hosted VOIP bill, the federal, state, and local taxes and fees run about 30% of the monthly billed amount. I live in Los Angeles, CA.) Can I expect that the taxes and fees for SIPStation will run about the same?
Posts: 2
Participants: 2
@draccusfly wrote:
I have a strange issue whereby an incoming international SIP trunk seems to be changing the incoming number from the SIP number to a landline number on the PBX, as a result it is routing incorrectly.
when the call comes in i see 22 lines ofAdded extension 'SIP NUMBER' priority 1 to ext-did-002 (these number priority 1 - 11 but each line is duplicated).
The next line shows:
Accepting call from 'SIP NUMBER' to 'LANDLINE NUMBER' on channel 0/6, span 1 (again this line is duplicated).
From then it routes based on the 'LANDLINE NUMBER'.
Testing with a different trunk results in the correct routing and I see
Executing ['SIP NUMBER@from-trunk-sip-provider:1] Set ("SIP/provider", "GROUP() =OUT_7" in new stack.
And the call correctly reaches its destination. What I can't figure out is why the first trunks is "converting" the SIP number into our main incoming line number
Drac
Posts: 2
Participants: 1
@mike_b wrote:
The other day I tried to call and got an "All channels busy" message.
When I checked the FreePBX dashboard, I saw that between 50 an 80
channels were busy. I called my SIP provider, and they told me that they
could only see one channel busy (the one we were talking on), yet me
dashboard continued to show me many busy channels.
I started the
Asterisk CLI and found a lot of activity there. It seems that there is
at least one rogue party out there that is probing Asterisk setups for
vulnerabilities. I have "allowguest" set to "No", and from what I
got from the SIP provider, there is no real calling going on, but it is
annoying to find out that someone is trying to break the system. Here is
what I typically see:-- Executing[971046406820677@from-sip-external:1] NoOp("SIP/myIP-00001aa9",
"Received incoming SIP connection from unknown peer to 971046406820677")
in new stack
-- Executing [971046406820677@from-sip-external:2] Set("SIP/myIP-00001aa9", "DID=971046406820677") in new stack
-- Executing [971046406820677@from-sip-external:3] Goto("SIP/myIP-00001aa9", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/myIP-00001aa9", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/myIP-00001aa9", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2015-06-29 16:10:59.841 MDT.
-- Executing [s@from-sip-external:6] Answer("SIP/myIP-00001aa9", "") in new stack
== Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/myIP-00001aa9'
-- Executing [h@from-sip-external:1] Hangup("SIP/myIP-00001aa9", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/myIP-00001aa9'(replaced my IP address with "myIP").
Is there any danger here? Is there a way to stop this completely? As it is
right now, I am getting one of these every couple of minutes or so.
Posts: 5
Participants: 2
@avayax wrote:
Is it possible to change the outgoing CallerID on my SIPSTATION trunk?
I am using SIPstation for outbound and another service for inbound, where my main company number is, which I would like to display on outgoing calls as well.
Posts: 2
Participants: 2
@avayax wrote:
Is there a feature where I can configure a forward if busy number on SIPStation trunks?
Let's say I receive an incoming call and all my trunks are busy, the call would forward to an alternate number.
Posts: 4
Participants: 2
@agasasterisk wrote:
Hi Experts,
I am unable get incoming calls from another phone system which does not register with USername or passwords. The Asterisk system is able to make outgoing calls to the same system. But for an incoming call, it wants the other system to be authenticated. I have got all the settings required for no authentication, but still it seems to be not helping. Anyone who would be able to help me with this ?
Trunk Configuration :-
host=20.1.1.170
type=friend
port=5060
insecure=invite,port
allow=ulaw
disallow=all
context=from-trunk
Even after these settings, I get the following message
<--- SIP read from UDP:20.1.1.170:5060 --->
INVITE sip:2723@20.1.1.58:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3477e7a810237
From: ;tag=704850~4ab333c0-314e-1172-16a8-eca8c1530263-31438422
To:
Date: Thu, 03 Sep 2015 19:24:58 GMT
Call-ID: 756d6680-5e819e8a-2f3dd-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: 1970103936-0000065536-0000002259-2852192532
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: 198v=0
o=CiscoSystemsCCM-SIP 704850 1 IN IP4 20.1.1.170
s=SIP Call
c=IN IP4 20.1.1.170
t=0 0
m=audio 25332 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 (NAT)
Sending to 20.1.1.170:5060 (NAT)
Using INVITE request as basis request - 756d6680-5e819e8a-2f3dd-aa010114@20.1.1.170
Found peer '2723' for '2723' from 20.1.1.170:5060<--- Reliably Transmitting (NAT) to 20.1.1.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3477e7a810237;received=20.1.1.170;rport=5060
From: ;tag=704850~4ab333c0-314e-1172-16a8-eca8c1530263-31438422
To: ;tag=as521e3e81
Call-ID: 756d6680-5e819e8a-2f3dd-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
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4be878a6"
Content-Length: 0
Any help on this would be great !
Thanks,
Posts: 10
Participants: 3
