7960 won't register 08.07.2009 12:02

Forum: Terminal Equipment

Hi all,

I'm having a bit of a problem with two of my new phones. I'm wanting to connect 3 phones behind a nat to pbxes. The siemens s450ip connects without any problems. However the sip cisco 7960 phones just won't register. I have tried everything to my knowledge; turned nat on the phones on, used a different pbxes server on each phone, using different ports. All 3 phones are able to connect to a Voipcheap account and make phonecalls. Even when the siemens phone that does connect is removed from the network, the cisco phones still won't register, they never have!

I'd appreciate any suggestions!

Servers down! 07.04.2009 03:03

Forum: Bugs

Hi,

Server www5 is down. Nothing works, phones won't register, I cannot place a call and the webbrowser gets stuck on "If you are not automatically redirected, please clear cookies or click here to access the forum".

Also, when I could still login earlier today I saw that all call recordings dating from before april 7th were not visible. So either someone from PBX made a mistake or someone is accessing my account without my consent and deleting my call recordings. I'm pretty sure the latter is not the case so I would appreciate some quick action from the PBX managers on both these issues.

Thanking you in anticipation,

outbound routing when forwarding is set in extension page 06.12.2008 18:15

Forum: Miscellaneous

Hi,

Can someone please tell me how to setup outbound routing so a forwarded call will not automatically go through the "default outbound route", but rather can be tied to the extension for which automatic forwarding has been activated.

many thanks!

EDIT: already solved it by adding the number used for forwarding to the outbound route in the 'Custom Dial Patterns' box. For some reason it doesn't work when you put it in the 'number starting by' box.

PBXes and Apple Iphone 08.10.2008 15:18

Forum: Terminal Equipment

Hi!

Anyone figured out a usable way yet to get pbx to work with an iphone (3g). I have been playing around with Fring which works but it won't run in the background so I always have to leave the program open in order to receive a call. I guess the only alternative would be to jailbreak the thing. Please post any thoughts ideas or experiences you may have with PBX (or voip in general) and the Iphone. Thanks!

Changing user name 15.06.2008 16:00

Forum: Miscellaneous

Hi,

I have tested my pbxes for personal use and would now like to introduce it into our corporate phone system. I need some step by step guidance to either change the user name or set up a new premium account and terminate my current subscription. i want ensure not to run into problems like I have read about on this forum involving the PBX system recognizing your account as a fraud. There will be extension IP's connected to the new account that have previously been connected to the current account. Also the new account will share some of the trunks that have been connected to this account. Again, the trunks or Ip's will not be connected simultaneously, I have no need for this account once the new account has been set up. Of course it would have my preference it it were possible to change the user name.

Thanks!

11.06.2008 00:33

Forum: Terminal Equipment

Hi!

Thanks again! I currently do not have any other UA's here at home. I tried replicating the issue with a professional SIP phone from work but it worked perfectly. So can you recommend me a good UA to replace the Siemens that offers reasonable value? Or do you have suggestions to make the Siemens phone work properly using PBXes? It might be something worth looking in to? I can hardly imagine Siemens shipped out millions of broken SIP phones, right? Thanks!

I will add the log of call that i just placed to GOOG 411 (took out some id's that might contain passwords or personal data (not sure they do, just wanted to be sure)). The behavior was exactly as described. That is, I did not get connected, but rather the Siemens phone produced the weird tone. Do you see any abnormalities?

Jun 11 00:34:25 VERBOSE[13221] logger.c: Found RTP audio format 0
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found RTP audio format 101
Jun 11 00:34:25 VERBOSE[13221] logger.c: Peer audio RTP is at port [homeIP]:26996
Jun 11 00:34:25 VERBOSE[13221] logger.c: Peer video RTP is at port [homeIP]:65535
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found description format PCMU
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found description format telephone-event
Jun 11 00:34:25 VERBOSE[13221] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Jun 11 00:34:25 VERBOSE[13221] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jun 11 00:34:25 VERBOSE[2831] logger.c: We're at 91.121.136.13 port 42170
Jun 11 00:34:25 VERBOSE[2831] logger.c: Video is at 91.121.136.13 port 41834
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x8 (alaw) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x10 (g726) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x400 (ilbc) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: -- Called JP_VoIP.ms/8004664411
Jun 11 00:34:25 VERBOSE[13221] chan_sip.c: SIP response 407 to standard invite
Jun 11 00:34:25 VERBOSE[13221] logger.c: We're at 91.121.136.13 port 42170
Jun 11 00:34:25 VERBOSE[13221] logger.c: Video is at 91.121.136.13 port 41834
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x8 (alaw) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x10 (g726) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x400 (ilbc) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 11 00:34:25 VERBOSE[13221] chan_sip.c: SIP response 100 to standard invite
Jun 11 00:34:39 VERBOSE[2831] chan_sip.c: Hangup call SIP/JP_VoIP.ms-325b, SIP callid [I D @ SERVER . COM]
Jun 11 00:34:39 VERBOSE[2831] chan_sip.c: Hangup call SIP/[A C C O U N T]-0001-f4c0, SIP callid [I D]@[HOME_IP]

07.06.2008 23:23

Forum: Terminal Equipment

Thanks for your response!

You're right a Siemens phone is producing the tone. I'd assume it does that because the server is not setting up the connection promptly, right? That is, when I replicate the behavior on a Softphone, I do not hear the tone, but rather I hear nothing until the connection is setup. Then again, sometimes the connection is setup straight away but the person I call cannot hear me until like 10 seconds after they've picked up. Like I said, sometimes when I get someone's voicemail, I do not hear anything until halfway through the greeting.

I experience these issues when I dial any number. I do not have the impression it is country related. Mostly however, I use voip to call people in the US. The account is hosted on the www5 server. I changed it to www1 to see if that changed anything, I can now report that it doesn't. I don't recall having these problems when I was on the Frankfurt server, but you guys closed that one on customers using 2gb+ a month. In retrospect, I do kinda think that was a bad move, especially without any advance notice or anything. I'm not saying this particular problem is related to that though!

Thanks!

Several annoyances 07.06.2008 20:57

Forum: Terminal Equipment

Hi,

I am experiencing two problems that i can't seem to be able to resolve. I have tried multiple outbound providers.

First when I place an outbound call I will first hear a tone (sometimes for as much as 40 seconds!) before I get connected. The tone I think is produced by my phone to signal the connection is not being set up. Then sometimes when the tone ends and the call is set up, I start out in the middle of the voicemail greeting of the person I'm calling.

Second, I get complains from people that I leave them 1 minute long SILENT voicemail messages. Then after some research I found out that I did not even leave those people a VM message and hung up before getting to the tone. When testing this by calling one of my own DID's and monitoring the STATUS screen I noticed that when I hang up the timer on the status screen keeps running, sometimes for as much as another 90 seconds!!!! Also when I place a call to a regular pstn line the phone keeps ringing way after I hang up the phone.

I have tested and confirmed this behavior on several phones, several extension, different termination providers and from different client ISP's.

Thanks for your response

13.05.2008 18:26

Forum: Bugs

Thanks for your reply! Actually i hadn't. But that would not explain why two incoming calls on two separate extensions is not working, right?

calls diverted to voicemail; "{name} is on the phone" 13.05.2008 12:49

Forum: Bugs

Hi,

I'm having a problem with my pbxes premium account. Whenever there is an external call on any of my trunks, I cannot receive a phone call on another trunk. This happens also when the calls are placed / received from different extensions. It seems as though my account does not allow for multiple simultaneous calls. Caller waiting with two calls on the same extension is not working either. As the subject line suggest, any secondary call is on the system is diverted to the extension's VM which answers by "{name} is on the phone".

I have checked my configuration and cannot find any error on my part which might trigger this behavior. I have done a 'submit and start' many times, that doesn't help.

Thanks!

VM notifications sender setting 03.05.2008 11:49

Forum: Feature Requests

Hi,

I was just wondering if it were possible to implement a function which lets you set the 'from' address for the VM notifications or maybe even edit actual text of the emails itself all together. I think for one it would make PBXes a lot more professional. I have still not implemented PBXes into our corporate phone system and this is one of the reasons why...

BR

VM notifications not being sent 02.05.2008 11:55

Forum: Bugs

Hi again,

I'm experiencing a problem with the Paris based WWW5 server. It's not sending out the email + voice attachments for voice mail. Or there might be a huge delay? I have tried both my own email server and Gmail's.

Thanks!

01.05.2008 19:43

Forum: Bugs

Hi,

Thanks for your quick reply. I can do a trace route, that is not the problem. The server www0 (frankfurt) has disappeared from the selection menu, so I cannot move my account to the server. My account was running on www0 until yesterday when this got changed without any notice. Mind you, I did not change any settings myself in the personal menu.

BR

Server down? 01.05.2008 16:56

Forum: Bugs

What happened to the server in Frankfurt Germany because it is gone from the selection list? Also I was until now not able to log in or make phone calls for almost 24 hours!

Language settings w/ 2 budgetphone accounts 12.04.2008 14:06

Forum: Bugs

Hi all,

I have been using PBXES for a while now and would like to take the opportunity to thank you for a great service. I have been experiencing a problem since I took out a second budgetphone.nl (dutch DID provider) account.

I have set up 2 trunks each registering to a separate account on sip.budgetphone.nl. I used the settings below as has been discussed on this forum many times before:

username: phonenumber@budgetphone.nl
password: password
sip server: sip.budgetphone.nl
The trunks are randomly named with a different name for each trunk.

Both budgetphone.nl accounts work and the appropriate phones rings whenever I place a call to either of the DID numbers. Regardless of which of the two numbers receives the call the "Call monitor" will in both cases show "sip.budgetphone.nl".

The problem starts when I select a DIFFERENT language in the trunk option pane for each trunk. When I do so the language settings do not seem to be affected unless I select THE SAME new language for BOTH trunks, which is not what I want.

Any help would be greatly appreciated!

Two more questions... 03.01.2008 20:30

Forum: Miscellaneous

Hi, I have two more questions. As you probably can tell by my previous post I'm pretty annoyed with your low bandwidth quota, but I saw that your other plan, PBXes PRO, does not such have a quota. When I tried to order it, however, it asked me for the additional number of extensions, minimum being 5. Does this mean that the minimum monthly charge for PBXes Pro is 5 times 9.95 EUR? I only need it for myself and occasionally my brother's Sip phone, but I think I use more than 5000 minutes a month on average. Also, I wanted to ask you whether it is possible to hold ONE free account along with a paid account, registered to the same name and IP address.

"Up to 2 voice channels" and 10gb bandwidth usage 03.01.2008 01:42

Forum: Miscellaneous

I signed up for a free account a while ago and have now upgraded to a premium account. Using the free account, any second incoming call would go straight to my VM, which would say something like "extension <number> is on the phone". The PBXes homepage reads that the free account comes with Up to 2 voice channels, which implies that under certain conditions you do not have two voice channels at your disposal. Could someone please tell me what those exact conditions are? All is working fine now that I have upgraded to the premium features.

Further, I was wondering if it’s possible to test for latency (jitter) b/w your servers and a third party VoIP provider. If not, then I would like to submit a request for such a feature to be added to your already impressive list of features. I think it would really help people in making a selection of VoIP providers that they want to use.

Also, I have noticed that the usage monitor equals every minute to 2 MB of usage. I read on this forum that even if you go as little as 500 MB over this limit you will be forced to pay 10 euro's (15 USD!) extra for that month even if your usage for the next month stays within the quota. I’d say that the softcap of 5000 minutes is a little unsatisfactory (only 2.5 hours a day, origination and termination combined!) and quite frankly unnecessary, considering that an extra GB of bandwidth costs only 0.09 EUR at Hetzner where the Nurnberg server is hosted. I understand that this service offers a whole lot more than just bandwidth availability, but one would assume that you already pay for this added value with your monthly subscription charge. Again, I love this website’s impressive set of features but I really think that bandwidth policy is a little ridiculous and ought to be communicated to the consumers more clearly (E.g. this service comes with 5000 minutes as we assume every minute of calling time to use 2 mb of bandwidth). Anyway, just wanted to put in my two cents on that! I’d appreciate a response though!

Thanks so much!