 | | Stupid question about voicemail Hi !
Ok, so I'm pretty new at voip.ms but so far I'm having fun looking at all the options and planning things I want to do.
Now, I do have a very stupid question. I want to add a personalized message to a voicemail. I know you can upload recording (and I did) and I want to play it instead of the default voice. Any way to do this ? (pretty sure there is, but I haven't found it yet).
Thanks ! | |
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 | DaveL
join:2005-11-12
Urbana, IL | Re: My Windows VoIP.ms monitor Gagdet This is a neat, informative gadget. Good job.
suggestions:
I would like to see normal time rather than military time.
I would like also to be able to reset/zero out the CDR length from time to time if possible.
--
7000s, 99W, 1370, 77sig, pro plan
"Notice how clear the skies were after 9/11 due to lack of con trails?"
»www.koehlerinjection.com | |
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| Reviews:
 ·voip.ms
| Hi, finally got a chance to install this little gadget but can't seem to get it to work.
It picks up the account # properly (though I do have two accounts a main and a sub account, I don't know if that makes a difference) but not the server or the amount remaining in the account.
I set the IP address on the voip.ms website to include both the IP address reported by my ISP *&* the address of my router, but no results are displayed.
I have an API password set and the API email I set to the email associated with the voip.ms account.
Any suggestions as to what I missed?
NefCanuck | |
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DaveL
join:2005-11-12
Urbana, IL | Separate phones Just getting voip.ms set up and wondering how to cable a second phone line.
My business is next to my house. All incoming lines are in this building with an underground conduit carrying internet and landlines to the house. I am wondering how to add another PAP2t at the house end...or if I even need to.
Here is the layout.
Linksys E2500 > biz pap2t and computers
Same e2500.> cable to house computers > ?
Another pap2t or something else or nothing else?
Hope I am clear on my question. Just puzzling through how to have internet and phone at the house end (with a different phone #).
Thanks
--
7000s, 99W, 1370, 77sig, pro plan
"Notice how clear the skies were after 9/11 due to lack of con trails?"
»www.koehlerinjection.com | |
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| PX EliezerPremium
join:2008-08-09
Hutt River
kudos:13 | PS: Virtually no one reads this sub-forum.
You really should post in the main VoIP Tech Chat forum.
»VOIP Tech Chat
This is a general VoIP issue, not provider-specific! | |
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| Reviews:
·voip.ms
| I'm not too sure about what you want to do, but if all the cables are connected to one line, it's very simple to do.
Identify the cable that goes to your office and disconnect it from the others in the demarcation box (the Bell box). Then you have two options, the easiest one will be to wire a new cable from the ATA to that existing cable and connect them with marrettes (or even better, if that cable has enough length, just connect it to a telephone jack next to your ATA), and of course connect that new cable to Line 2 of the ATA.
Option 2 if you cannot install a new cable, take the existing cable near the ATA and identify which one it is in the demarc box. You will see it's either an old telephone cable with 2 pairs of twisted wires (Green/Red and Black/Yellow) or a new CAT5e cable with 4 pairs of twisted wires (Blue/White should already been used, and three other unused pairs).
Then near the ATA you will buy a new telephone plate with two jacks. The top one you will connect the red/green (or blue/white) cable, and the bottom jack you will connect the black/yellow (or orange/white) in the red/green spot. Then connect the black/yellow cable to the office cable (black connects to green and yellow connects to red).
That's it! You then connect Line 2 to Jack #2 (bottom) and you got your second line working.
If you're lost with the colours, here's a quick summary on the CAT5 conversion:
Green = White with blue lines
Red = Blue with white lines
Black = white with orange lines
Yellow = orange with white lines
And if you connect the black/yellow to the jack, again make sure you connect black cable to green screw, and yellow cable to red screw. If you use one of those levitron jacks, follow the code sticker but remember that you are really connecting a green/red despite the actual cable colour!!!
Let me know if you're lost and give us details of your current cabling. | |
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Reviews:
·voip.ms
 ·TekSavvy Cable
| Unknown CID on outgoing calls Recently I have been told that my CID displays Unknown, Private Number or Unavailable to some of my recipients. I opened up a ticket with voip.ms and was advised to test premium route. I tried and it has fixed the issue. The only problem is that now my bill goes from $10 to $16 a month and I don't want to be paying $6 extra just for guaranteed CID on outgoing calls.
Is there any other way to fix this without the Premium route?
Does freephoneline have this issue as well?
I primarily switched over to voip half a year ago from traditional land line for the sole reason of saving money. I initially was going to go with freephone line but decided to give voip.ms a try after hearing so much about it. With premium route, I will be paying half the bill of what I used to pay with traditional land line ($16 as opposed to $32).
I am also experiencing dropped calls, not hearing ring tones on outgoing calls until they answer / goes to their voice mail and some other odd issues. Now I understand this could be due to my router but I am in this predicament now. Should I go back to land line, try freephoneline or how do I fix this voip.ms issues without having to pay $16 a month.
Thank you! | |
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| | Reviews:
·voip.ms
·TekSavvy Cable
| Re: Unknown CID on outgoing calls Thanks to both of you. I am using PAP2T-NA and have it tuned already. I've done everything possible with the ATA when it comes to tuning. I believe the problem lies in my D-LINK DIR-625 router. I have also done everything possible to tune that. It's definitely time for a router upgrade.
As you already know, the CID issue has nothing to do with my end. Like I said, I don't want to be paying $6 extra just to be guaranteed my outbound CID. I will post in the general voip forum.
Thanks. | |
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| PX EliezerPremium
join:2008-08-09
Hutt River
kudos:13 | PS: Almost no one reads this sub-forum.
You'd get much more response in the main VoIP forum.
»VOIP Tech Chat | |
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| Reviews:
·voip.ms
| I have similar issues with dropped calls, ringtones, etc, but I pay on average $3.50/month so I don't really care. I guess I could try the premium route to see if they would go away, considering I barely spend any money on this, paying an extra dollar a month would not hurt me. | |
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| Reviews:
·voip.ms
·TekSavvy Cable
| Thanks for the feedback guys. I am thinking about getting FPL's unlock key and have both FPL and voip.ms at home. I just have a question here.
I will be connecting a 4 handset panasonic cordless phone system to FPL through PAP2T-NA. I will use the second line on PAP2T-NA for voip.ms using a separate cordless phone system (vtech). I want both the panasonic (FPL) and vtech (voip.ms) cordless phones to ring at the same time when I receive calls at either number. How do I do this? Thanks!
I'll be basically making outgoing calls using FPL and will be receiving calls mainly on my voip.ms (flat rate incoming). The reason why I don't want to get rid of my voip.ms account is because I want to keep that number and if in the future I really like FPL, then I might port it over. | |
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| | Reviews:
·voip.ms
| Re: Unknown CID on outgoing calls said by erfans:I will be connecting a 4 handset panasonic cordless phone system to FPL through PAP2T-NA. I will use the second line on PAP2T-NA for voip.ms using a separate cordless phone system (vtech). I want both the panasonic (FPL) and vtech (voip.ms) cordless phones to ring at the same time when I receive calls at either number. How do I do this? Thanks!
I'll be basically making outgoing calls using FPL and will be receiving calls mainly on my voip.ms (flat rate incoming). The reason why I don't want to get rid of my voip.ms account is because I want to keep that number and if in the future I really like FPL, then I might port it over.
I don't think it's possible, as far as I know the PAP2T-NA will not "link" both lines, it would have to be done with the provider, but in this case, you cannot make a ring group since you'll be using two providers.
The best solution would be to get an ATA that would allow you to have two different accounts, one for outgoing, one for incoming, on the same line. The SPA-3102 does that if I'm not mistaken, and I have read that you can install its firmware on the PAP2T, but don't quote me on this, you will have to do some research and if you want to do it, you might ruin your ATA. | |
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 JEDI
join:2005-04-11
Longueuil
kudos:1
 ·ELECTRONICBOX
·Videotron
| No ring when calling people with SPA112 I am new to voip and so far it is working great. I am using a Cisco SPA112 and it is working fine. I read some blogs, the voip.ms wiki and some forums to do some tweaks.
There is someting I don't like much though. It is not a major issue and doesn't prevent the service from working but when I call someone I don't hear any ring while waiting for them to answer. At first I though it didn't work (I though my call was not going through) but as I said the calls go through. Is there a setting in my voip.ms account or my Cisco SPA112 that I need to enable to hear the ring when I am making a call? | |
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| JEDI
join:2005-04-11
Longueuil
kudos:1 | Re: No ring when calling people with SPA112 It looks like I was mistaken since I hear ringing now but I didn't change anything. Anyway, no problem now  | |
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| | | | Re: No ring when calling people with SPA112 I've had this happen maybe once. Didn't change anything and it never happened again after that. Usually the closest thing I have is that it sometimes takes several seconds before I hear the other phone ring when I call someone. | |
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1 edit | It happens from time to time, same thing when my parents with Bell call me, sometimes it doesn't ring and I pick up, but I heard 2-3 rings.
Edit: I have a PAP2-TA | |
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|  xsbell
join:2008-12-22
Canada
kudos:1
·Primus Telecommu..
1 edit | Are you sure it's working fine?
Can I ask which firmware version you have on your adapter?
I have nothing but problems with this POS since I first bought it. I have the newest firmware (011) installed and it loses registration 10+ times a day, then eventually crashes and needs to be reset every week, or two. (known bug, check the cisco forum)
Jul 22 08:39:09 SPA112 kern.warning [17179607.604000] #### RTP STOP Flag set in this channel break ####
Jul 22 08:39:09 SPA112 kern.warning [17179607.616000] ###### sock_sendmsg return 172
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 2 and str G.726/8000 chan 0
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 18 and str G.729/8000 chan 0
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Jul 22 08:39:09 SPA112 kern.warning [17179607.624000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Jul 22 08:39:14 SPA112 daemon.notice msgswitchd[183]: MSGSWD RTCP Reqt len 12 Data 2,2009268,7304,0
Jul 22 08:39:14 SPA112 kern.warning [17179612.312000] RTCP is running so calling rtcp stop
Jul 22 08:39:14 SPA112 kern.warning [17179612.312000] chan->kmode is present not null
Jul 22 08:39:14 SPA112 kern.warning [17179612.316000] ###### RTCP sock_sendmsg return 172
Jul 22 08:39:14 SPA112 kern.warning [17179612.340000] ###### sock_sendmsg return 172
Jul 22 08:39:14 SPA112 kern.warning [17179612.340000]
Jul 22 08:39:14 SPA112 kern.warning [17179612.340000] #### RTP STOP Flag set in this channel break ####
Jul 22 08:43:38 SPA112 syslog.notice syslog-ng[120]: STATS: dropped 0
Jul 22 08:53:38 SPA112 syslog.notice syslog-ng[120]: STATS: dropped 0
Jul 22 09:03:38 SPA112 syslog.notice syslog-ng[120]: STATS: dropped 0
Jul 22 09:13:39 SPA112 syslog.notice syslog-ng[120]: STATS: dropped 0
There is supposedly beta firmware available that fixes this issue, along with the Voice Mail indicator fix, but you have to contact them to open a case and then they'll give you a link to the firmware. | |
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| | JEDI
join:2005-04-11
Longueuil
kudos:1 Reviews:
·ELECTRONICBOX
·Videotron
| Re: No ring when calling people with SPA112 Nope unfortunately I spoke too quickly and it doesn't work. I am quite dispointed because it seems really hard to get it to work while I assume Cisco would mean quality.
I also have the latest firmware. I tried to get the beta but it is difficult to get. They will want you to send logs. I tried to talk to tech support and got bored of waiting. | |
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| | | I'm having the same issue and I have found what I think is a solution but a potential bug on these devices.
I am running their latest firmware 1.1.0 (011) Feb 10 2012. Go into your line configuration and check your network jitter level. set this to high or extreme and it will resolve your calling out/ no ringing issue.
Now to produce the error above, set the network jitter level to low and make sure the Jitter Buffer Adjustment is set to yes. Submit changes and now try dialling out. Chances are you are going to get dead air. Now hang up and set the jiiter level to extreme high and call out, you will now get a ringing with out any issues.
Whats even weirder is that if you set the Jitter Buffer Adjustment to No and set the level to low, you won't get a ringing on the line when dial out. Set to high and viola, everything works even when you have use jitter to no.
I have reported this case to Cisco just 30 mins ago. Thier support wants me to run a packet capture with wireshark and send them my logs.
Setting the jitter level to Medium will cause the device to sometimes ring, like i experienced this morning. | |
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| pmicho
join:2012-08-24
Montreal, QC Reviews:
·Acanac
·ELECTRONICBOX
| Hi,
I have same problem with spa2102. It seems like it mostly happens with calls to cell phones. Is that possible? I have tried playing with network jitter level, but I get the same problem with Jitter level to Very high.
Same problem with pap2 at my parents.
Dont know what to do. | |
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| |  Incoming Caller Id Not Working, Broken Ring
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| Reviews:
·Start Communicat..
·voip.ms
·Rogers Hi-Speed
| Re: Incoming Caller Id Not Working, Broken Ring This forum is a bit hard to find, so you may want to post this in the VoIP Tech Chat forum: »VOIP Tech Chat Lots of knowledgeable people there. I'd suggest adding the make/model of your ATA in your post. I've never had this problem with voip.ms, so I have to idea (I'm using a PAP2T). | |
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| | Newbie having trouble with incoming calls I have trouble with cell signals in my house and my Android cell phone is constantly searching for a signal killing my battery. So I decided to try and use voip for my cell phone calls over wifi. I got a voip.ms account as well as an iNum and installed cSipSimple client on my phone and it seemed to register fine. I can make outgoing calls using the cSipSimple client but I cant figure out how to get incoming calls. When I call the 0118335100XXXXXXXX iNum number from a land line nothing seems to happen. What do I need to do to get my cell phone to ring? In cSipSimple I have enabled Use WifFi for Incoming calls and Always Available under Availability Profile. I apologize in advance if this is a stupid question but I am a newbie and any help would be appreciated. | |
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| xsbell
join:2008-12-22
Canada
kudos:1 Reviews:
·Primus Telecommu..
| Re: Newbie having trouble with incoming calls said by abrahavt:I have trouble with cell signals in my house and my Android cell phone is constantly searching for a signal killing my battery. So I decided to try and use voip for my cell phone calls over wifi. I got a voip.ms account as well as an iNum and installed cSipSimple client on my phone and it seemed to register fine. I can make outgoing calls using the cSipSimple client but I cant figure out how to get incoming calls. When I call the 0118335100XXXXXXXX iNum number from a land line nothing seems to happen. What do I need to do to get my cell phone to ring? In cSipSimple I have enabled Use WifFi for Incoming calls and Always Available under Availability Profile. I apologize in advance if this is a stupid question but I am a newbie and any help would be appreciated.
I believe cSipSimple only has the speex and G722 codecs set by default, which VOIP.ms doesn't support.
I have mine set to use GSM and PCMU/PCMA and I don't have any problems. | |
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| | Time to Leave voip.ms? Over the past months, their reliability has gotten worse. While they respond to support cases quickly, about all support can say is to try another server.
It seems they never fix their connections issue. Today, I was starting to get cross connected calls.
I'm looking for a new vendor. | |
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See 7 replies to this post |
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Reviews:
 ·Bell Fibe
·ELECTRONICBOX
| Transferring my number to VOIP questions While I am still pretty green in the whole VOIP scene. I setup my adapter, purchased the a test DID and everything seems to be working great.
I want to transfer my main line over, pretty much because I do not want to lose the number but my home phone is so rarely used that it is silly to pay the amount I do each month for it.
A few remaining questions
- Is there a FAQ on what the proper process for this is? When my line is transferred I want to just kill the DID I purchased and make the new number my main line with the voicemail attached. I have a horrible fear of accidentally killing the number I have had for the last 20 years :P
- I'm topping up $25 at a time with Paypal, once this balance is reached it just stops the line until a new payment is received? (i.e nobody could hack my line and run up a huge LD bill?). I know it will advise me at $10 but I want to make sure I cannot run up a huge credit
thx! | |
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| | | | Re: Transferring my number to VOIP questions Thanks for the replies, it is in fact for voip.ms
I'm having a new ISP installed tomorrow and if there are no issues I will start the DID portability process
cheers | |
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| | voip.ms voicemail blinking light Hi, anybody knows how setup my granstream ht702 or voip.ms in order to have my phone light blinking when a have a voicemail. I tried a few things and did'nt work.
thanks
Guildor | |
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| | | Re: voip.ms voicemail blinking light said by Guildor :Hi, anybody knows how setup my granstream ht702 or voip.ms in order to have my phone light blinking when a have a voicemail.
Here is the manual.
»www.grandstream.com/products/ht_···lish.pdf
"Disable Visual MWI" should be set to NO.
"Subscribe for MWI" should be set to YES. | |
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| | "No service configured error" When dialing out My ata device (OBi110) was working just fine, but suddenly it was unable to receive any incoming calls or dialing out. I tried a few things to finaly find out that it was the cable between the device and the phone.
One of the thing I did, was to reset the device to the default configuration. I reconfigured it as explained on the voip.ms wiki site. But since I did it, I've got the "No service configured error" where ever I try to dial out. Incoming calls are working fine.
Any idea ?? | |
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| | | Re: "No service configured error" When dialing out Solved!
The parameter : Physical Interfaces -> PHONE Port -> PrimaryLine was set to PSTN line by default. Just had to set it to SP1 and everything is working now. | |
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| | Auto-number input on callback? I'm using Android Jelly Bean. This is my current method when voip.ms calls me back.
1. I get a callback.
2. I use the dial pad to make an outgoing call. (Which means I have to memorize the number I am calling before making the call).
Is there a way for Android to make a call by scrolling through the name list or inputting it?
This might sound lazy but I don't want to memorize numbers. | |
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| Dee BeePremium
join:2005-05-08
North York, ON | Re: Auto-number input on callback? This method is not an auto method but you can use *75xx feature to quick dial frequently called numbers which is set up by using the phone book feature. | |
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| | voip.ms + DISA + Android dialer I use voip.ms and have a DISA line setup for calling when I'm out of Wi-Fi and have to rely on cell phone minutes. Problem is that I'd like to find a way to automate the whole process of calling my DISA access number and punching in the password and the number I ultimately want to call, i.e., I'd like it to work in conjunction with my address book. Anyone found a solution to this issue? Thanks. | |
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| | | Re: voip.ms + DISA + Android dialer I have this exact setup. I use an Android dialer called "prefixer", it's the best out there as far as I know.
»www.253below.com/prefixer | |
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| | | | Re: voip.ms + DISA + Android dialer I've installed that program but haven't been able to get it running.
Do you set rule as when number matches " .* " ?
Also, is there a code or symbol you use to force it to actually dial up the DISA before it punches in the password?
I assume use commas for pause, but not sure if that triggers a dial command as well or if you need a different code.
Thanks. | |
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| | | Re: voip.ms + DISA/callback + IP SIM dialer Better... if you're using a SIM card device:
interweb search: IP SIM dialer
A film circuit is inserted between the SIM and socket adding the seamless functionality of disa/callback dialing.
No phonebook masochism required! _This_ functionality is completely legal. There are shady uses for such tech. | |
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My Windows VoIP.ms monitor Gagdet
