[Aviator168]

Originally Posted by rikisphinx

Yes my pbx has a public ip.

When i take out the ip address in the username as you recommended it just does't reach the * server. Any idea?

Normally, you don't need an IP address as part of the user name. Maybe your real user name in the provider is with the IP address, but I doubt it.

[ricgar]

Originally Posted by Aviator168

When you do the SIP debug in the SIP(Asterisk), can you tell where(IP address and port) the response was send to? It looks like both BB and server are in the same subnet and you are using WIFI.

here is the response from the * debug

<-- SIP read from 194.168.2.103:50408:
REGISTER sip:sip.org SIP/2.0

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=1234567890;rport

Max-Forwards: 70

Contact: <sip:aviator168@192.168.0.1:5060;rinstance=1234567 890>;expires=0

To: "12345"<sip:aviator168@sip.org>

From: "12345"<sip:aviator168@sip.org>;tag=a25e2f7e

Call-ID: 1234567890

CSeq: 1 REGISTER

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Length: 0


---(10 headers 0 lines) ---

Using latest REGISTER request as basis request
Sending to 194.168.0.1:5060 (NAT)
Transmitting (NAT) to 194.168.2.103:50408:
SIP/2.0 404 Not found

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=1234567890;received=194.16 8.2.103;rport=50408

From: "12345"<sip:aviator168@sip.org>;tag=a25e2f7e

To: "12345"<sip:aviator168@sip.org>;tag=as59065f3b

Call-ID: 1234567890

CSeq: 1 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0


---
June 3 19:20:50 NOTICE(51444): chan_sip.c:11328 handle_request_register: Registration from '"12345" <sip:aviator168@sip.org>' failed for '194.168.2.103' - ACL error (permit/deny) Scheduling destruction of call '1234567890' in 15000ms

Destroying call '1234567890'

confusing!

[ixlwqwmqtx]

Originally Posted by Aviator168

1. I just tried the website blackvoib.webs.com. It is working fine.

That is very strange...I have tried a number of times throughout the day and get "HTTP 500 - Internal Server Error" every time. Tried again just now and still not able to reach the site.

EDIT: I VPNed to another network and tried... I was then able to access your website. So the problem seems to be with my home broadband connection for some odd reason.

[ixlwqwmqtx]

Originally Posted by ixlwqwmqtx

That is very strange...I have tried a number of times throughout the day and get "HTTP 500 - Internal Server Error" every time. Tried again just now and still not able to reach the site.

EDIT: I VPNed to another network and tried... I was then able to access your website. So the problem seems to be with my home broadband connection for some odd reason.

When I run a tracert on the server name I get:

C:\WINDOWS>tracert blackvoib.webs.com

Tracing route to blackvoib.webs.com [216.52.115.50]
over a maximum of 30 hops:

  1    <1 ms     1 ms    <1 ms  DELETED
  2     2 ms     2 ms     2 ms  DELETED
  3    47 ms    43 ms    44 ms  DELETED
  4    39 ms    40 ms    39 ms  DELETED
  5    41 ms    41 ms    40 ms  DELETED
  6    49 ms    47 ms    45 ms  DELETED
  7    41 ms    44 ms    39 ms  DELETED
  8    49 ms    39 ms    40 ms  DELETED
  9    40 ms    39 ms    40 ms  px0.bllon.uk.easynet.net [212.134.155.194]
 10    41 ms    41 ms    41 ms  ge0-0-3.ar8.bllon.uk.easynet.net [212.134.155.209]
 11    40 ms    40 ms    39 ms  195.50.122.121
 12    42 ms    47 ms    39 ms  ae-34-52.ebr2.London1.Level3.net [4.69.139.97]
 13   111 ms   109 ms   110 ms  ae-42-42.ebr1.NewYork1.Level3.net [4.69.137.70]
 14   127 ms   125 ms   126 ms  ae-3-3.ebr4.Washington1.Level3.net [4.69.132.93]
 15   116 ms   122 ms   125 ms  ae-74-74.csw2.Washington1.Level3.net [4.69.134.182]
 16   119 ms   117 ms   115 ms  ae-2-79.edge2.Washington4.Level3.net [4.68.17.83]
 17   115 ms   115 ms   119 ms  COX-COMMUNI.edge2.Washington4.Level3.net [4.53.114.58]
 18   116 ms   146 ms   118 ms  ashbbbrj02-ae0.0.r2.as.cox.net [68.1.0.221]
 19   117 ms     *      117 ms  border7.te2-2-bbnet2.wdc002.pnap.net [216.52.127.87]
 20     *        *        *     Request timed out.
 21     *        *        *     Request timed out.
 22     *        *        *     Request timed out.
 23     *        *        *     Request timed out.
 24     *        *        *     Request timed out.
 25     *        *        *     Request timed out.
 26     *        *        *     Request timed out.
 27     *        *        *     Request timed out.
 28     *        *        *     Request timed out.
 29     *        *        *     Request timed out.
 30     *        *        *     Request timed out.

Trace complete.

C:\WINDOWS>

where I have "DELETED" the first several entries. I am not sure what this all means but was suggested on a forum. It appears it may have to do with my ISP restricting access to blackvoib.webs.com. Does anyone have any suggestions?

Originally Posted by Aviator168

3. It might be an issue with 9700. Our team is investigating this model.

I also have an old BB 9000 which I will install BlackVoib on once I am able to access your website via my home broadband.

[ixlwqwmqtx]

Ah ha. The download link on the BlackVoib website links to:
http 173.220.21.106:9999/VoIP/BlackVoib.jad

I can access this through my home broadband connection even though my ISP seems to be blocking blackvoib.webs.com.

[ixlwqwmqtx]

(3) I can not hear anything on the speaker of my device during a call but the microphone is working.

Originally Posted by Aviator168

3. It might be an issue with 9700. Our team is investigating this model.

I have installed v0.52 on my BB 9000. When I make a call I am not able to hear any ringing through the handset. I am not able to get the audio to work on either of my BB 9000 or BB 9700.

EDIT: It is working on the BB9000 now (after a reboot). But the audio is very loud and, as you said before, it is not possible to adjust the audio. It is too loud to be very usable for anything but testing.
I installed v0.52 on by BB9700 but the audio still is not working on that device.

[rikisphinx]

Originally Posted by Aviator168

What do you mean routable ip address? Do you mean your pbx has a public ip address?

Aviator,
When i took out what you asked me to take out two things happenned : the first thing is that on the phone display i get the message : " Network start succeeded .... " and after that "NR idle" is displayed. From that point if i try to make a call it says " calling...." but nothing else happen.

From the *console even when i have ' Network start succeeded ' i still have that error :

chan_sip.c:15699 handle_request_register: Registration from '"12345"<sip:aviator168@sip.org>' failed for '166.192.112.194' - No matching peer found

I don't know that user 12345 and as i already told you my user ID is 2222.
IP address of the registrar cannot be resolve

When i use the stun server, it says that it cannot resolve the registrar server.

AS i told you my pbx is using an public IP.
Thanks

[Aviator168]

Originally Posted by rikisphinx

Aviator,
When i took out what you asked me to take out two things happenned : the first thing is that on the phone display i get the message : " Network start succeeded .... " and after that "NR idle" is displayed. From that point if i try to make a call it says " calling...." but nothing else happen.

From the *console even when i have ' Network start succeeded ' i still have that error :

chan_sip.c:15699 handle_request_register: Registration from '"12345"<sip:aviator168@sip.org>' failed for '166.192.112.194' - No matching peer found

I don't know that user 12345 and as i already told you my user ID is 2222.
IP address of the registrar cannot be resolve

When i use the stun server, it says that it cannot resolve the registrar server.

AS i told you my pbx is using an public IP.
Thanks

Can you PM me an extension so I can try it here? It is very strange to me as you are using an Asterisk server.

[Aviator168]

Originally Posted by ricgar

here is the response from the * debug



confusing!

I looks like there is firewall between your BB device and the SIP server, and your firewall is blocking the UDPs from the SIP getting to your BB device.

How do I know?
<-- SIP read from 194.168.2.103:50408:

Transmitting (NAT) to 194.168.2.103:50408:

That 50408 port should be 5060 which is assigned to BlackVoib on the device. I suggest you take a look at your network.

[ricgar]

Originally Posted by Aviator168

I looks like there is firewall between your BB device and the SIP server, and your firewall is blocking the UDPs from the SIP getting to your BB device.

How do I know?
<-- SIP read from 194.168.2.103:50408:

Transmitting (NAT) to 194.168.2.103:50408:

That 50408 port should be 5060 which is assigned to BlackVoib on the device. I suggest you take a look at your network.

Hmmmm, i did notice that, but the funny thing is that i took a BB9700 and placed the same exact setting on the same network and that reistered fine and was working in both directions, but my storm 2 still does the same thing, what could that be?

But there is no firewall, just my dsl router (wireless) which connects all the machines & the two BB, odd!!

[Aviator168]

Originally Posted by ricgar

Hmmmm, i did notice that, but the funny thing is that i took a BB9700 and placed the same exact setting on the same network and that reistered fine and was working in both directions, but my storm 2 still does the same thing, what could that be?

But there is no firewall, just my dsl router (wireless) which connects all the machines & the two BB, odd!!

I think something is up with your router. Eventhrough we have not test it on any Storm and we are not going to at least for the moment, it should not behave like that.

[lmcjipo]

I just downloaded the program (v0.52) from the website and configured it.

I tested it with my carrier's 3G data network on my Blackberry Bold 9700 and the call quality is what I would consider to be good. There is a delay and according to the person that I called, he also hears his own voice (echo) a few seconds later. I'm not sure if him hearing his own voice is because his voice was coming from the speakerphone or because of the program itself. In terms of the latency/delay, I felt that it wasn't too bad. It was noticeable so I wouldn't use it for a long call but for a short call, I think that it is okay. I talked for roughly 1 minute and 30 seconds over the carrier's 3G data network and didn't get disconnected.

All in all, I think that it is a very good program.

What I would like to see implemented are:
1) integration with the Blackberry contact/addressbook so that I can either select the contact from within your program or from within the Blackberry contact/addressbook, I can press the "menu" button and one of the options would be to use your program to make the call
2) have the sound come from the Blackberry earpiece instead of the speakerphone (or better yet, have it as a user toggle-able item where the user can pick whether he/she wants the sound to come from the speakerphone or the earpiece).

That's all the comes to my mind right now based on my ~2 minutes of testing.

I will try it over WiFi tomorrow morning.

[ricgar]

Originally Posted by Aviator168

I think something is up with your router. Eventhrough we have not test it on any Storm and we are not going to at least for the moment, it should not behave like that.

Tried this now on a completely different network, and it still does the name thing! i think it is a problem then with the BB storm 2's and the software. I shall then wait for this to be resolved.

Great job thus far on this work though!

If there is any assistance i could offer you guys do let me know!

[ixlwqwmqtx]

I am still having problems with calls always being disconnected after 20s (and occasionally with one-way audio) and was referred to this help page by my SIP provider:
www sipgate.co.uk/faq/index.php?do=displayArticle&article=602&id=249
Both issues are mentioned in the FAQ.

My device registers on the Sipgate network and I can place a call. Two-way audio sometimes works. I am wondering if it is an issue with ports not being forwarded on my router. Is the STUN server or UPnP supposed to solve this problem automatically? I could try to set my device to a static IP address and open the ports on my router but I would like it to work without having to hack the router settings b/c eventually I will want to use the program from public wifi spots which I will not necessarily have access to.

I suppose my main "wish list" enhancement for this application is to have access to a log file so I can see what/where the problems are and to provide more info to the developers.

Thanks.

[Aviator168]

Originally Posted by ixlwqwmqtx

I am still having problems with calls always being disconnected after 20s (and occasionally with one-way audio) and was referred to this help page by my SIP provider:
www sipgate.co.uk/faq/index.php?do=displayArticle&article=602&id=249
Both issues are mentioned in the FAQ.

My device registers on the Sipgate network and I can place a call. Two-way audio sometimes works. I am wondering if it is an issue with ports not being forwarded on my router. Is the STUN server or UPnP supposed to solve this problem automatically? I could try to set my device to a static IP address and open the ports on my router but I would like it to work without having to hack the router settings b/c eventually I will want to use the program from public wifi spots which I will not necessarily have access to.

I suppose my main "wish list" enhancement for this application is to have access to a log file so I can see what/where the problems are and to provide more info to the developers.

Thanks.

If you are behind a router and you are using a SIP service provider, you should always specify a STUN server even some service providers say it is optional. Otherwise, you will only get oneway audio at best.

[gravyflex]

Tried this now on a completely different network, and it still does the name thing! i think it is a problem then with the BB storm 2's and the software. I shall then wait for this to be resolved.

I have the same issue with the 8520 (Curve 2)

[lmcjipo]

I did some more testing with my Bold 9700 over both WiFi and my carrier's 3G network and there is a serious echo problem according to the people that I called. Also, when I'm speaking with them and they are a few feet/metres away from me, I notice how bad the latency/delay is. It is ~3 seconds.

I'm not sure if the echo problem could be due to their voice coming out of the speakerphone (and them hearing their own voice via the mouthpiece when it is "broadcasted" over the speakerphone). I attempted to lower the volume with the volume controls on my Bold 9700 and there doesn't seem to be any volume control with the program.

For the sound quality received by the caller/user, I would rank it as good. For the sound quality according to the receiver, I would rank it is either poor or fair. For the latency, I would also rank it poor or fair.

I probably wouldn't have an issue using the program for short calls (under 30 seconds) but anything longer would probably be too annoying to the person I'm calling (because of the echo and latency issues) and to me (mainly latency issue).

I have used my iPhone 3GS with another SIP VoIP program and the echo is hardly noticeable to the other person so the problem doesn't appear to be provider/carrier/SIPserver related but seems to be software related.

[themlruts]

Originally Posted by Aviator168

Did you specify a STUN server?

No stun server
This quoted from call centric site

What is the address for Callcentric's STUN server?

Callcentric has developed its NAT traversal technology without using STUN. This technology is user agent independent. If you need information on how to configure your user agent, please visit the support section.

THanks
Mike

[themlruts]

also getting network start failed on both wifi and 3g

[Aviator168]

Originally Posted by themlruts

No stun server
This quoted from call centric site

What is the address for Callcentric's STUN server?

Callcentric has developed its NAT traversal technology without using STUN. This technology is user agent independent. If you need information on how to configure your user agent, please visit the support section.

THanks
Mike

You can use any STUN server.

Two here

stunserver.org
stun.xten.com

I went to their support -> setup page. They want you to open the ports on the router which requires firewall configuration.