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8
Posted by
FreePBX Developer
2 months ago
Stickied post

The old FreeSwitch Wiki is dead!

Unfortunately, when you go to wiki.freeswitch.org/anything it's redirecting you to freeswitch.com without any hints as to where to go, rather than (In my opinion) the correct place of https://freeswitch.org/confluence/display/FREESWITCH

I expect we'll have a bunch of people going 'The FreeSwitch wiki is broken!' so, to head that off, please go to https://freeswitch.org/confluence/display/FREESWITCH instead and then search inside confluence for what you want.

Sorry 8-(

4

Does this seem like a good deal?

I won an auction I forgot I bidded on. The price seemed good but just today I found out I won it; no one else bidded on it.

It is this.

Polycom VVX300 phones. The POE injectors I am ordering cost more lol.

Looking for what others have to say. Thanks

Edit: Updated correct link

4

Mobile VoIP carrier with support for SMS Short Codes

I recently migrated my personal Canadian cellphone number to Fongo. I've found it decent: the app works well, far cheaper than a second cell plan (I use it on my work phone). The one thing I'm missing is support for SMS short codes, which are used often in 2-factor authentication.

Fongo say they are working on short code support, however that seems to have been the status quo for years so I'm not too hopeful there. Is there a mobile voip service that does support short codes? I do see that jmp.chat supports sms & short codes but that is not a voip service and there is no ability to receive phone calls.

Edit: Should have mentioned, I'm in Canada so unfortunately Google Voice is not an option.

Edit2: noted jmp.chat actually would support phonecalls to voicemail or via a SIP client... that's looking to be better and better. Alternatively I may just set up a jmp.chat number just for receiving my 2-factor auth stuff.

3

Choosing a solution for the next 6-7 years

Hi VOIP ! We are in the process of choosing a new partner for our telephone system (23 sites, 727 telephones) We have 9 Avaya IP office 500 v2 right now and we want to simplify our telephony. We have basic telephony needs + 2 call centers.

We are seriously thinking about getting rid of Avaya. The new server edition based they quoted us is very expensive and we have to buy new licences again.

We have a quote from a great VOIP partner that we trust who work directly with Shoretel (Mitel). The pricing and the demo met our expectations and we are impress with the system for our need.

We also met a solution provider who build their own system on a Netsapiens base and are using Polycom generic phones. They have lot of big customer in our area and their system and price also met our needs.

We don't know what's going to happen with the Shoretel/Mitel fusion and we don't know if Netsapiens have any future. Avaya seem big and expensive and their support/license profile is bloating the price.

I am looking for your expertise, help and knowledge about these system and your future view of the telephony world.

Thank you

4

Need a new cloud based phone system, and recommendations?

I’ve been using eVoice for many years without a problem, but I am getting much busier and about to inherit a large new client list and eVoice won’t be able to handle my needs for reasons I’ll explain at the end.

Necessities: - Unlimited minutes - Auto Attendant - Call Forwarding without requirement of an app to receive calls (effectively, must be able to ring a landline) - Cloud based (we are completely mobile, using cell phones - Call Queueing - Customizable hold music - Ability to leave a voicemail at any time during hold - Multiple phone lines (at least 3) - Multiple extensions (at least 6)

Would like to have: - Text message forwarding - Transcribed Voicemails - Fax receiving

Preferably: - less than or around $50/month - something scalable

Can anybody recommend anything that may work for me? I’ve done a lot of research but still don’t know why one is better than another and looking for some advise.

——

Why eVoice won’t work: - The largest plan is 4000 min, or 2.2 hours a day. I am on the phone more than that every day - Call queuing is not an option. People don’t like leaving voicemails but often I’m on the other line. I would like for them to be able to leave a voicemail if they would like or wait for me to be available if they’d prefer

2

Looking for voip that supports fax

I'm wondering if anyone knows 100% if magicjack works with sending and receiving faxes. If not is there any that does? I'm in canada and we pay 100 a month for a land line that's essentially only there for faxing.

0

Would with power cable work with this phone?

I have purchased Polycom VVX 310 phone off of ebay. It does not come with power plugs. At the time I was considering purchasing, I thought that I would have more than enough power cords to use but I just want to make sure I have the right one so I don't risk damaging the phone once I get it.

This plug says 100-120V 43-67Hz 0.6A for input

For output it says, 12V

All of the OEM plugs for the Polycoms say 48V for ouput.

Should this work because I see no reason why it shouldn't.

Thanks

3

Now that Simonics doesn't work, what should I use for calling via my Polycom VVX400?

I was using the cheap Simonics service, what can replace it for little to no cost? Just need outbound US only calling.

2

MicroSIP - can't place outgoing calls (408 or 503 error) - voip.ms

Hey,

I was wondering if anyone has had experience with this. I'm running MicroSIP on windows 10 and I'm unable to make outgoing calls. they terminate with error 408 or 503.

I followed their troubleshooter on the website. I don't have a SIP proxy, my login is fine (shows online and I'm able to receive calls) I've tried public STUN servers and I've tried with and without allo IP rewrite.

I chatted in with voip.ms and they didn't have a solution

The application is allowed through the windows firewall. I was able to my calls to work with Zoiper so I might have to go back to that.

Thanks!

0

Best way to route calls based on callerid from a database of numbers

We're looking for a way to route incoming calls based on a large database of numbers to the appropriate office location by their caller ID. We're not against 3rd party options or first routing calls to another service prior to 3cx if that's required.

Currently running a cloud hosted 3cx system but looking to try out other options so something that pre routes and forwards to different did's would allow for Pbx changes and be preferred.

8

Looking for VOIP Provider for local number in Trinidad

I live in Canada, and I have a number of relatives in Trinidad. I'm trying to find a VOIP provider in Trinidad that can provide a DID so I can have a local incoming number. I've got my own FreePBX server at home, so I don't need hosted service. Hoping to keep it cheap - so far I've only found providers around $45/month minimum.

  • I've contacted TSTT who apparently have SIP trunk services, but there's not a lot of information on their systems or pricing.

  • vol.co.tt sells PBX services, not VOIP services.

  • Sent an email to Digicell TT to see what they provide. They have Cloud PBX and SIP Trunking services for business.

I haven't found much else. Can anyone recommend any VOIP providers local to Trinidad?

1

Is there a way to convert rj11/DECT phone signal to something else?

My internet modem comes with an rj11 jack that allows you to attach a DECT handset, and my subscription includes free landline phone service.

Before my provider was taken over by a bigger one, they used to offer an Android app that you could use as an alternative access method to make VOIP calls using the same landline number, so I didn't need to have a charged DECT phone lying around unnecessarily. Unfortunately, the app is not supported anymore (the new provider owns a mobile network.)

Are there any adapters or converters that would allow me to make phone calls on other devices than a DECT phone, for example by converting the DECT phone signal to a USB, Micro USB (OTG) or Bluetooth connection for Android or Windows?

The modem itself is locked and I cannot change the firmware or any phone-related settings.

4

ISP VoIP, are they always so difficult?

So I got a contract for full IT at a local business. Looking at replacing the ISP phone system with my own. As luck would have it my switch doesn't support LLDP-MED which the ISP uses to assign vlans. I requested that the phones just statically be assigned to respective vlan, issue solved. OMG this was apparently the wrong thing to ask, they just went and threw the contract at me, got all pissed I was upgrading stuff, bare in mind they do nothing for IT support, strictly phones. They are 100% unwilling to assist or cooperate to the benefit of our mutual client. All this does is reaffirm the need to replace the ISP's phone system. I dont need a partner that unwilling to support a client.

3

Static on lines - Call Manager -- new issue

In the past month, most of my users are getting really bad static on calls to/from external numbers only. The challenge in troubleshooting is that our gateway shares 3 PRIs with another company, and they aren't having any issues.

Only my users are able to hear the static.

Any guesses as to whats up? Thanks in Advance

3

[Switchvox] Calls coming in from a door intercom to only ring certain extensions?

Hey all. Question here about something that seems like it should be fairly straightforward, but isn't immediately obvious and was hoping to pick the brains of people more knowledgeable about this than I.

They are using Switchvox, and this relates to a previous post I had in here about their door intercom. What they want to have happen is when someone rings that intercom, they only want certain extension to ring, and not all extensions which currently happens. I feel like this is a matter of making an extension group of the extensions its supposed to ring, and then making an IVR that rings that extension group when the intercom is used, but the IVR only has the option to ring a specific extension, not a group. Is this something that stacks? That if I make rules for multiple extensions it will ring multiple extensions? Am I even looking down the correct path for this? Thanks for reading, and for any suggestions!

3

How to set up inbound calls for high availability/redundancy

I know this is kind of a broad question but does anyone have any suggestions for how to set up a number so that calls can be delivered 99.999% of the time? Outbound is easy - we can just set up multiple ISPs, redundant SIP trunks, etc, but what about inbound? I can't come up with any way to eliminate a single point of failure.

We have it set up currently with a local carrier with fiber ran into the building. We have good call quality, the phones generally still ring when the internet is down, and we really haven't had many problems before today, when the ISPs session border controller failed and it took them 7 hours to come out and replace it. I have used VOIP.ms in the past but not enough that I can tell the C levels that porting the number to the cloud is going to actually reduce outages in the long term.

5

Remote site with analog, need gateway that can do 'failover'

I have a customer with multiple sites. We're using SIP at their main site to their Switchvox system. Their remote sites have a VPN tunnel and the phones connect to the system over that. The problem we're trying to solve is that they have a site in a small rural town where they cannot port the number to a SIP carrier. So they have the old analog phone and also have a SIP phone. My thought was to eliminate that with an analog gateway and connect that back to the system. That's easy enough to do, but I'd also like the IP phone there to be registered to the gateway. Then if the gateway can't place a call to the system it can use some type of failover rules or a hunt group to call the registration on the local IP phone. Since this is a small town internet can be spotty and thus why I want to have it set up this way. Anyone done anything similar?

6

Google Cloud restrictions?

I was looking at replacing our office VOIP provider with 3CX. They offer speedy setup through Google Cloud, but I just saw through bvoip the restrictions they have:

(e) unless otherwise set forth in the Service Specific Terms, use the Services to operate or enable any telecommunications service or in connection with any Application that allows Customer End Users to place calls or to receive calls from any public switched telephone network;

Why would 3CX offer Google cloud as a partner to setup service if Google's restrictions specifically say no PTSN connections?

I already went through the setup to test it out, but I'm concerned that Google may close my account if I continue using their cloud services. Also, does AWS have the same restrictions? Or who would be a good hosting provider?

10

Testing e911 in Canada

It was quite difficult to track down official information on testing e911 service in Canada, so I followed the generic instructions.

I phoned my local RCMP to get the 911 centre's emergency number

The guy there was confused and asked that I throw away the number the RCMP gave me, as it was not for the general public but said that he'd let everyone on shift know I'd be calling. He needed my name, number and address.

I called 911 from my device and was connected to someone at a completely different call centre who had no idea what I was talking about. Apparently there is a daily testing code you can get for the national emergency response centre by following the instructions here : http://www.northern911.com/911-services/official-testing/

They said they understood I was testing, but were required to transfer me to emergency services anyways as I didn't have the code. Of course I'm glad they're strict in this regard.

I was transferred thru 2 more levels of call centres before reaching my regional one. The national operator said please transfer to police, but the caller is just testing his system

To which the regional person said, "Oh, is this DoctorPhish? It's working!"

So I didn't have police show up at my door, but if I'd known about the daily testing code it could've saved me a tense couple of minutes

7

Polycom microbrowser status screen nagios status and time

I wrote up a quick php script to show the current number services with each status type and put a 24h clock in the corner of the screen. Thought it might be useful for someone.

<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
<html>
<head><title></title></head>
<body>
Nagios Stats --------- <?PHP print(date("H:i")); ?><br/>
<?PHP

$PARSESTATUSDAT="cat /var/cache/nagios3/status.dat | grep current_state= | sort | uniq -c | sort -n | awk '$1=$1' | sed 's/current_state=0/Services OK<br\/>/g' | sed 's/1 current_state=1/1 Service Warning<br\/>/g' | sed 's/1 current_state=2/1 Service Critical<br\/>/g' | sed 's/current_state=1/Services Warning<br\/>/g' |  sed 's/current_state=2/Services Critical<br\/>/g' | sed 's/1 current_state=3/1 Service Unknown<br\/>/g' |  sed 's/current_state=3/Services Unknown<br\/>/g'";

system($PARSESTATUSDAT, $retval);

?>
</body>
</html>
7
comment
3

Setting up phone system for small business

I am looking to set up a phone system for my small business. We currently have 2 lines in the US and 7 lines physically in India. The India lines use United States numbers because all incoming/outgoing calls are United States. What would be the best option to go?

  • Shoretel/Mitel

  • Vitel

  • LiveOps

  • Monmouth

  • Vonage

  • Sonetel

or if you have any other recommendations it would be greatly appreciated!

Requirements:

  • make calls in and out of India using United States numbers

  • be able to call from a phone or computer

  • have one service number that customers/clients call then dial an extension

2

SipXCom can see polycom phones but can't seem to send / recv config or SIP traffic from them?

I'm trying to setup a stupidly simple set of ip phones in my apt and well it's been a bit of an up-hill battle, even though I'm a developer by trade. My issue is that I've been really struggling to figure out how to properly provision polycom phones / get them to actually respond to jobs from SipXCom.

Right now, SipXCom can see the devices, firware version, etc. However, whenever I send a new profile or even a reset request the requests just time-out.

I've also tried FreePBX but the PHP app always breaks whenever I get to the configurator to send Polycom profiles.

I'm honestly about to give up on this project because the VOIP protocol is so unnecessarily convoluted.

2
0

which voip has the best UI? still looking for options to replace gvoice

so never used voip before, but guessing its like gvoice

that you can use it on the web and mobile etc and it has a UI to

  • send texts
  • & make calls

there's various problems with the gvoice UI, too many to detail, all of which could be considered marginal tho

also missing some UI-related features

quesiton: any detailed video anayslsis to find this?


seems that gvoice has stopped working with many 'voip' devices or something (dunno anything about voip so dunno what they talking about)

source: http://goedhartvoordieren.nl/?page=VOIP/comments/8iw76v/june_18_google_poised_to_blow_up_google_voice/


question also do we know what kind of isp is good for voip? seems that satellite internet is bad for voip


it seems according to a comment on http://goedhartvoordieren.nl/?page=VOIP/comments/8vopiq/need_the_a_better_voip_alternative_to_google/

that most voip has all kinds of problems, question: so maybe voip is not ready for prime time? if so ill keep looking once a year i guess

3

Hdlp with Voip phone for sweet hard of hearing mom in memory care

Hi. My mom's memory Care unit requires VoIP phones. My brother installed a Cisco SPA301 1 line IP phone and it's not doing the trick. Mom can't hear it ring and can't hear me if she does see the light and pick up, when I do get through to her.

I call Mom every night and now that we've moved her to this facility I've gone from speaking with her every to only getting through maybe once a week.

Her Dr. Diagnosed her with moderate to severe hearing loss in left ear and moderate hearing loss in right ear this week.

She is 86.

She has dementia and is highly functional but we need to be careful of anything at all that could be a trip and fall hazard such as long cords.

I've downselextrd to three phones for mom but none are VoIP. How can I make them compatible with VoIP and not have a. A big footprint (the table she uses for her phone is about 12 inches in diameter and she doesn't want wall mount) and not complicated because I am not tech savvy with VoIP.

The phone must retain the handset volume between use (current one does not) and must be easy to use. The family photos offered on some phones are not necessary but may be nice. Also, must have a small footprint (she wants a table versus wall mount) and not be a trip fall hazard.

The models I am considering are assistech.com's picture phone; clearsounds csc 500 amplified spirit phone and the Cadillac so to speak, krown HD 65 amplified telephone.

I think the facility uses AT&T ...I don't know why but I get a 'the wireless customer you are trying to reach is unavailable" error message all the time. Which never made sense to me.

Oddly no other residents have a similar problem because I visit weekly and have asked the staff hoping they could help.

I can try to post links to the phones I'm researching and I do have pictures of mom's current crappy phone if that would help.

Any help in a. Suggesting phones and b. Teaching me how to convert an non VoIP phone to Voip, simply and easily and safely for mom are greatly appreciated.

Most articles are over my head on this topic. My work phone is VoIP and I'm familiar with it but that's my only experience.

Thanks so much in advance.

TLDR: nearly deaf mom with dementia needs reliable VoIP phone in memory care facility.

2

pjsip outbound call issue (inbound works)

**EDIT*\*

Solved and the solution is at the end of the post

Hi all,

I have a private voip server for keep myself in touch with my relatives. I would like to move from the current vps provider to a new one for better service/location/etc.

The current setup is a FreePBX (chan_sip) configuration that I would like to swap to native Asterisk 13 and pjsip. I have a trunk as well.

I was able to (manually) migrate the users into the new environment, we are able to call each other. Fortunately I was able to register my trunk and I can receive call from outside, but I cannot call any external number. I followed this documentation: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-ExampleSIPTrunkConfiguration

I have only one dialplan for test as well as I use tcpdump for catch the packets.

Please take a look on the details, hopefully you will find the source of the issue, why I get "SIP/2.0 403 Invalid From username" error message.

This is my trunk and endpoint for the trunk setup

;=============== TRUNK
[ephone]
type=registration
transport=simpletrans
outbound_auth=ephone_auth
server_uri=sip:<trunkproviderurl>:<not5060port>
client_uri=sip:<trunkproviderurl>:<not5060port>
retry_interval=60
line=yes
endpoint=ephone

[ephone_auth]
type=auth
auth_type=userpass
password=<trunkpassword>
username=<trunkusername>

[ephone]
type=endpoint
context=from-external
disallow=all
allow=ulaw
allow=alaw
allow=gsm
outbound_auth=ephone_auth
aors=ephone

[ephone]
type=aor
;max_contacts=1
contact=sip:<trunkproviderurl>:<not5060port

[ephone]
type=identify
endpoint=ephone
match=<trunkproviderurl>

This is my dialplan setup:

[internal]
exten => <hardcodedmobilenumber>,1,Answer()
same => n,NoOp("${EXTEN} is on the way")
same => n,Goto(to-external,1,1)

[from-external]
exten => <landlinenumberforinboundcallwhichworks>,1,NoOp("Call has been arrived, Ringing")
same => n,Ringing()
same => n,Dial(PJSIP/<oneendpoint>,7)
same => n,Hangup()

[to-external]
exten => 1,1,NoOp("Goto done")
same => n,Ringing()
same => n,Dial(PJSIP/<hardcodedmobilenumber>@ephone,10)
same => n,Hangup()

TCPDUMP error message

17:12:36.649710 IP (tos 0x10, ttl 58, id 0, offset 0, flags [DF], proto UDP (17), length 460)
    <trunkproviderurl>.<port> > <myserver>.sip: [udp sum ok] SIP, length: 432
    SIP/2.0 403 Invalid From username
    Via: SIP/2.0/UDP <myserverip>:5060;received=<myserverip>;rport=5060;branch=z9hG4bKPjcd0dea05-9b32-491b-bde5-7ed106dd310d
    From: <sip:<internalendpoint>@<myserverip>>;tag=10ee30f5-ff33-4e6a-8ba5-cd0d8257f6ae
    To: <sip:<hardcodedmobilenumber@<trunkproviderurl>>;tag=d3a0af37d228242dac7e07e7cd12f538-e702
    Call-ID: 9280ed2e-1608-4cb7-8b31-43f3cc12c02a
    CSeq: 8066 INVITE
    Server: MESE driven Ephone SIP system
    Content-Length: 0

The "old" sip setup is (FreePBX not cli configured)

register=<username>:<password>@<trunkproviderurl>:<not5060port>/<username>

[ephone_out]
username=<username>
type=peer
secret=<password>
port=<not5060port>
insecure=invite
host=<trunkproviderurl>
context=from-trunk-sip-ephone_out

[<username>]
username=<username>
type=user
secret=<password>
fromuser=<username>
fromdomain=<trunkproviderurl>
context=from-trunk-sip-ephone_out

Any help would be really appreciated.

I am happy to share any additional information if you need.

Thank you very much!

2

Epygi QX50 1 way audio only over Peplink VPN, Want 2 Way

Greetings, I recently installed an Epygi QX50 VOIP server in my office, with Yealink T48s phones. I have a Peplink Balance 305 router at the office and Peplink Balance One at home, which provides me a Peplink pepVPN between my home and office.

(TLDR - T48s user can hear, but no one can hear user)

I have attempted to use of my Yealink T48s phones at my home as an extension to the system. Everything works EXECEPT for Microphone audio from the the person using the phone. The phone user can hear the caller, but the mic audio of the user is not transmitted. (TLDR - T48s user can hear, but no one can hear user)

I have created rules in the peplink firewalls for incoming, outgoing and internal firewall, on both ends. I created a blanket rule to allow all traffic between the two networks. (I have tried the phone onsite at the office, and all audio works when connected to the same lan as the QX50)

In searching the Epygi forums, I found several similar older posts (5+ years old) that had issues with one-way audio. However, there was never any resolution posted.

Anyone with a similar experience, or have any ideas?

Thank you for reading.

2
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