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BT Problems

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Thank you so much for your time in explaining these things to me.

I put in the IP address of the server with the 5000/provisioning port details and it has downloaded the phone book etc, all those things we did not have before.

All our phones (only 4) are Grandstream GXP-1200

In the past all I had done was input the static ip and gatewey etc details and then went to the account 1 and input each extension into each phone maually and it worked.

These have worked like a charm for over a year.

In my recent move and the subsequent problems with the 3CX - phone connections I made the following changes:

Upgraded 3CX from v9 to V10
Upgraded the firmware of the phones to 1.2.5.3
Upgraded my ethernet switch but I have reinstated the old switch (has mde no difference)
Input the new 5000 port for the provisioning pathway and that has given the phones more bells and whistles...

however, I cannot for the life of me work out why the connection of phone to 3cx server is taking so long!! there is obviously an issue either at the phone end or 3CX end, i need to make more investigations...
 
Once connected (registered), does it stay connected?

Normally, when you reboot the phone, it will make an attempt to register immediately. In some cases, if the registration does not occur, the phone will make additional attempts rather quickly, but if these "fall back" attempts are not successful, then the phone may wait until the next registration period comes about. This could take some time depending on what the timeout period is programmed to be within the phone.

It seems somewhat encouraging that you are now able to get the phone book as this does imply connectivity.

Is it safe to assume that there is only one NIC installed in the 3CX system?
Have you tried to turn off the internal MS firewall and see if that improves the situation?
 
Once the phones connect they stay connected.

the firewall on the 3CX server was never turned on, since the server is a dedicated 3cx server I was taking a calculated risk to switch of inbuilt XP firewall.

I have noticed since the 5000 port change the phone does some nice things like let u know who u r calling but the call takes a long time to connect internally and now some progress I can ring out but get a cancel voice when it connects with BT...

However it takes 2/3 minutes to connect...

My log has something to report:

18:50:40.781 [CM503008]: Call(5): Call is terminated
18:50:37.515 Session 143 of leg C:5.1 is confirmed
18:50:37.406 [CM503007]: Call(5): Device joined: sip:[email protected]:40600;rinstance=a079ce2201f5f8e4
18:50:37.390 [CM503007]: Call(5): Device joined: sip:[email protected]:5060;transport=udp;user=phone
18:50:37.390 [MS210005] C:5.1:Answer provided. Connection(proxy mode):192.168.1.68:7018(7019)
18:50:37.375 [MS210001] C:5.5:Answer received. RTP connection[unsecure]: 127.0.0.1:40618(40619)
18:50:37.375 Remote SDP is set for legC:5.5
18:50:37.375 [CM503002]: Call(5): Alerting sip:[email protected]:40600;rinstance=a079ce2201f5f8e4
18:50:37.187 [CM503025]: Call(5): Calling Unknown:Ext.EndCall@[Dev:sip:[email protected]:40600;rinstance=a079ce2201f5f8e4]
18:50:37.187 [MS210004] C:5.5:Offer provided. Connection(proxy mode): 127.0.0.1:7020(7021)
18:50:37.171 [CM503016]: Call(5): Attempt to reach <sip:[email protected]:5060;user=phone> failed. Reason: Not Acceptable HereReason Unknown
18:50:37.156 [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 407 Proxy Authentication Required; from IP:62.239.15.136:5060
18:50:36.921 [CM503025]: Call(5): Calling VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.921 [MS210002] C:5.4:Offer provided. Connection(transcoding mode): 109.156.84.206:59331(59332)
18:50:36.859 [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 407 Proxy Authentication Required; from IP:62.239.15.136:5060
18:50:36.625 [CM503025]: Call(5): Calling VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.625 [MS210002] C:5.3:Offer provided. Connection(transcoding mode): 109.156.84.206:59329(59330)
18:50:36.546 [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 407 Proxy Authentication Required; from IP:62.239.15.136:5060
18:50:36.265 [CM503025]: Call(5): Calling VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.265 [MS210002] C:5.2:Offer provided. Connection(transcoding mode): 109.156.84.206:59327(59328)
18:50:36.171 [CM503004]: Call(5): Route 3: VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.171 [CM503004]: Call(5): Route 2: VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.171 [CM503004]: Call(5): Route 1: VoIPline:01213285010@(Ln.10002@BT)@[Dev:sip:[email protected]:5060]
18:50:36.171 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=phone>
18:50:36.171 [MS210000] C:5.1:Offer received. RTP connection: 192.168.1.27:5006(5007)
18:50:36.171 Remote SDP is set for legC:5.1
18:50:36.156 [CM505001]: Ext.100: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.5.3] PBX contact: [sip:[email protected]:5060]
18:50:36.156 [CM503001]: Call(5): Incoming call from Ext.100 to <sip:[email protected]:5060;user=phone>
18:50:36.140 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK655977a016b0916c
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp;user=phone>
To: <sip:[email protected]:5060;user=phone>
From: "Shokat Aziz"<sip:[email protected]:5060;user=phone>;tag=d846e58f7f6ca306
Call-ID: [email protected]
CSeq: 10501 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE, PRACK, MESSAGE
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:[email protected]:5060;user=phone",nonce="414d535c0607b66b28:368f64e4aadc3b4acd7039bbabe274a5",response="0ea91d74dc34f3262b8e48a10245df98"
Supported: replaces, timer, path
User-Agent: Grandstream GXP1200 1.2.5.3
Content-Length: 0
X-Grandstream-PBX: true
P-Early-Media: Supported

18:50:27.062 Currently active calls [none]
18:49:55.062 Currently active calls [none]
18:49:48.421 [CM503008]: Call(4): Call is terminated
18:49:40.515 Session 132 of leg C:4.1 is confirmed
18:49:40.390 [CM503007]: Call(4): Device joined: sip:[email protected]:40600;rinstance=a079ce2201f5f8e4
18:49:40.375 [CM503007]: Call(4): Device joined: sip:[email protected]:5060;transport=udp;user=phone
18:49:40.375 [MS210001] C:4.3:Answer received. RTP connection[unsecure]: 127.0.0.1:40616(40617)
18:49:40.375 Remote SDP is set for legC:4.3
18:49:40.359 [CM503002]: Call(4): Alerting sip:[email protected]:40600;rinstance=a079ce2201f5f8e4
18:49:40.187 [CM503025]: Call(4): Calling Unknown:Ext.EndCall@[Dev:sip:[email protected]:40600;rinstance=a079ce2201f5f8e4]
18:49:40.171 [MS210004] C:4.3:Offer provided. Connection(proxy mode): 127.0.0.1:7016(7017)
18:49:40.156 [CM503016]: Call(4): Attempt to reach <sip:[email protected]:5060;user=phone> failed. Reason: Temporarily Unavailable
18:49:40.156 [CM503003]: Call(4): Call to sip:[email protected]:5060 has failed; Cause: 480 Temporarily not available; from IP:77.72.169.134:5060
18:49:34.671 [MS210003] C:4.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.1.68:7014(7015)
18:49:34.671 [MS210001] C:4.2:Answer received. RTP connection[unsecure]: 77.72.168.215:38048(38049)
18:49:34.671 Remote SDP is set for legC:4.2
18:49:34.656 [CM505003]: Provider:[Freecall] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:5060]
18:49:34.656 [CM503002]: Call(4): Alerting sip:[email protected]:5060
18:49:33.812 [CM503025]: Call(4): Calling VoIPline:00441213285010@(Ln.10001@Freecall)@[Dev:sip:[email protected]:5060]
18:49:33.812 [MS210002] C:4.2:Offer provided. Connection(transcoding mode): 109.156.84.206:59325(59326)
18:49:33.718 [CM503004]: Call(4): Route 1: VoIPline:00441213285010@(Ln.10001@Freecall)@[Dev:sip:[email protected]:5060]
18:49:33.718 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=phone>
18:49:33.718 [MS210000] C:4.1:Offer received. RTP connection: 192.168.1.27:5102(5103)
18:49:33.718 Remote SDP is set for legC:4.1
18:49:33.703 [CM505001]: Ext.100: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.5.3] PBX contact: [sip:[email protected]:5060]
18:49:33.703 [CM503001]: Call(4): Incoming call from Ext.100 to <sip:[email protected]:5060;user=phone>
18:49:33.687 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK67869510b3276839
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp;user=phone>
To: <sip:[email protected]:5060;user=phone>
From: "Shokat Aziz"<sip:[email protected]:5060;user=phone>;tag=8417b883592fd62f
Call-ID: [email protected]
CSeq: 32073 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE, PRACK, MESSAGE
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:[email protected]:5060;user=phone",nonce="414d535c0607b62d54:ef5ae1caeb7ef2f46a8f9364e83f7095",response="c9ef031a0d4666cf36770e2a400cb0cc"
Supported: replaces, timer, path
User-Agent: Grandstream GXP1200 1.2.5.3
Content-Length: 0
X-Grandstream-PBX: true
P-Early-Media: Supported
 
The ogs show some issues with proxy authentication. You be to check your settings and credentials to ensure they are correct for your various providers.
 
As suggested by the experts on this forum I took the plunge and reinstalled XP pro including sp3 and the related updates and security patches.

I also installed windows installer 4.5.

I dowloaded a fresh copy of 3CX v10 and also v11 beta

I had of course backed up my 3CX system settings and DB and installed it my memory pen-drive.

4 hours later, I was ready to install 3CX. Everything was going well until the end when an error was reported, windows installer had a problem, a script failed to run. Then 3CX uninstalled itself and I cannot get ito go any further.

Then I tried to install v11 and had the same problems.

I did not expect any installation issues and am stuck and need answers quickly. Please help...
 
This is the screen shot
 

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What net framework versions are installed?
 
You of course realize -

Minimum Requirement

Windows XP will not be supported in the final release of 3CX Phone System Version 11. If you use Windows XP, we highly recommend that you upgrade to Windows 7 or Windows Server 2008 / 2008 R2 editions. For a full list of supported Windows operating systems,

If you intend to move to V11, then I encourage you to go ahead and move to a supported OS. Some have reported success with XP, but that is not a guarantee, nor does it mean that while it may work today that future updates and service packs may render it unworkable tomorrow.
 
Ineblett

Sorry I could not reply until today, back to work again.

I checked and have the following .Net:

V4.0 Client and V4.0 extended.

I checked in the registry and the add/remove programs utility.

From the information you have provided I will NOT install 3CX v11. I only did that in the 1st place so that I can get the damn thing working again.

I really do not want to reinstall XP again and with it the patches etc which will take another 4 hours.

However, I do wish to tell you that before I had installed SP3 and .net 4 and installer 4.5 I did run the 3cx v10 a couple of times and hence found the hard way that I still needed the patches and the upgrades, maybe this premature installation is nowing causing me so many issues?

As alway I look forward to your help in this matter.
 
OK, some good news I have been able to install 3CX v10 on my new build xp.

The problem was the .net version was too new and after installing version 3.5 the install sailed through.

Unfortunately I never had a backup of the working BT setup, and I made changes to the old setup thinking that BT was at fault when it was the internal setup...

I have ran the configuration api and have taken the settings from the url address bar including the password which has been included in the 3CX voip provider setup page.

I cannot connect to BT and still get the cancel voice prompt when it goes through, however when making internal calls, it still takes an extraordinary amount of time to connect.

So any thoughts on these issues would be gratefullyy accepted.
 
More good news

BT is working again on 3CX, I have made a backup of it. Horaah!!

However, It takes 2/3 minutes to connect internally and externally...

so there must be a setting on 3cx or even the gxp-1200s that needs to be changed in order for them to ring and connect immediately.

Any ideas on this issue which I can't figure out; and is driving me nuts...
 
Well, unfortunately with the new install things have changed and we will need to start from the beginning. This time however, let's segment the issues so as to address one thing at a time. Let's start with the internal calls:

Can you post a log of a call from one extension to another?

Is it safe to assume that earlier, before these issues, that a connection between internal extensions was almost seemless; you hit dial/send and the other phone rang almost immediately? Have you made any changes to the gxp in any of your recent efforts to troubleshoot? What about the network? Can you confirm that the SIP Server address installed in the gxps is the internal private (192.168.x.x or 10.x.x.x) address of the system where you have 3cx installed?
 
In the past the phones used to call each other with a lad of about 5 seconds.

The following is a log from the 3CX server for:

an internal call from

ext 100 to ext 103

----------------------------------Server Log----------------------------------------------------

4:23:02.500 [CM503003]: Call(4): Call to sip:[email protected]:5060 has failed; Cause: 487 Request Cancelled; from IP:192.168.1.21:5060
14:23:02.375 [CM503008]: Call(4): Call is terminated
14:23:00.921 Currently active calls - 1: [4]
14:22:48.625 [CM505001]: Ext.103: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.5.3] PBX contact: [sip:[email protected]:5060]
14:22:48.625 [CM503002]: Call(4): Alerting sip:[email protected]:5060;transport=udp;user=phone
14:22:48.531 [CM503025]: Call(4): Calling Ext:Ext.103@[Dev:sip:[email protected]:5060;transport=udp;user=phone]
14:22:48.531 [MS210006] C:4.2:Offer provided. Connection(by pass mode): 192.168.1.27:5006(5007)
14:22:48.500 [CM503004]: Call(4): Route 1: Ext:Ext.103@[Dev:sip:[email protected]:5060;transport=udp;user=phone]
14:22:48.500 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=phone>
14:22:48.484 [MS210000] C:4.1:Offer received. RTP connection: 192.168.1.27:5006(5007)
14:22:48.484 Remote SDP is set for legC:4.1
14:22:48.484 [CM505001]: Ext.100: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.5.3] PBX contact: [sip:[email protected]:5060]
14:22:48.484 [CM503001]: Call(4): Incoming call from Ext.100 to <sip:[email protected]:5060;user=phone>
14:22:48.484 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bKc2b0a7c02455ab8c
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp;user=phone>
To: <sip:[email protected]:5060;user=phone>
From: "Caller Name"<sip:[email protected]:5060;user=phone>;tag=e75db39f2b8fa0d5
Call-ID: [email protected]
CSeq: 33791 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE, PRACK, MESSAGE
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:[email protected]:5060;user=phone",nonce="414d535c060b6c2889:f98301362d734f2db00e2e40f938420f",response="ce4895f1a4e5288213585620dbe60b50"
Supported: replaces, timer, path
User-Agent: Grandstream GXP1200 1.2.5.3
Content-Length: 0
X-Grandstream-PBX: true
P-Early-Media: Supported

------------------------------------------end of log------------------------------------

The server address on the GXP-1200 interface is the same as the IP of the 3CX

ie 192.168.1.68

Pls see attachment,

I hope this helps...
 

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From your post, it appears that the call was "handles" in less than a second, until it failed because of a problem with IP 192.168.1.21 a second later.

atomicwaste12 said:
4:23:02.500 [CM503003]: Call(4): Call to sip:[email protected]:5060 has failed; Cause: 487 Request Cancelled; from IP:192.168.1.21:5060

Which means that it is the phone, delaying the sending of the dialled number, or perhaps because of the failure to ring the extension (and no audio feedback/ringback tone?), it appears to be much longer.

Are the TCP/UDP options on the set, an either/or, or can you set to use both?
 
I don't know that it will make a difference, but you do not need to specify the port in the IP address. It should read the IP only 192.168.1.68. You also do not need to specify the outbound proxy, the field can be left blank altogether. The SIP port is already specified in the field further down. Do not know that it hurts being the way it is, but I have not had to specify and the Grandstreams I have work fine.

Leejor - with regard to the UDP/TCP question, it is one or the other...at least on the Grandstream phones I use (unfortunately not this model).
 
Hi Guys, thank you so much for your efforts in sorting this out. So in desperation I have done the insane thing and have posted the screen shots of the advance settings and account 1 settings, there are six shots!!!, sorry for taking up bandwidth, but I need all the help I can get, thank you.
 

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I do not see anything terribly obvious, but the disctintive ring tone settings I do not believe are correct. It should be looking for a specific number and not the orgination status. Remove the current settings.

Grandstream phones also support star codes - *XX. I was already in the midst of posting this when the idea hit me that perhaps a star code has been invoked unknowingly and that is why 103 is terminating the call unexpectedly. Almost like DND is in effect. Leejor is correct in that the call is being routed within a second, but in the log you posted it looks to me like the call will never go through for some reason.

It may be worth your whileot go to 103 and do a factory reset and then reprovision the phone from 3CX and then try the same test. Following are the instructions to do so -

INSTRUCTIONS FOR RESTORATION:
Step 1: Press “OK” button to bring up the keypad configuration menu, select “Config”, press “OK” to enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)
Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping:
0-9: 0-9
A: 22 (press the “2” key twice, “A” will show on the LCD)
B: 222
C: 2222
D: 33 (press the “3” key twice, “D” will show on the LCD)
E: 333
F: 3333
Example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to move the cursor or wait for 4 seconds to continue to key in another “2”.
Step 3: Press the “OK” button to move the cursor to “OK”. Press “OK” button again to confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.
 
I notice that you have No Key Entry Timeout set to 30 seconds, which is a bit excessive, as the default is 4 seconds. I'm assuming that you are using the octothorp (#) as the end of dialling each time?

Otherwise, you will wait 30 seconds for the digits to be sent to 3CX after the last key press.
 
Guys, you are brilliant, thank you so much for your help. Wow.

This was the issue, by pressing # after dialling a number rings through immediately. Phew.

I did change it to 30 secs because when I am looking at a new client number and am dialling it - it used to time out and had to key in the number again and again and everyone in the office was annoyed with this issue. How can I change this so that the phone rings immedaitely but I also get more time to key in my number too.

Thanks once again.
 
A number of SIP phones/ATA's, allow for an internal dialplan, I don't believe that yours is one of them. The dialplan allows you to create rules (much like the outbound rules in 3CX), until a rule is matched, the phone will wait for more digits. Once matched, the digits are sent out immediately. Cisco/Linksys devices, and others, make use of this.

You will either have to continue to use the #, or lower the setting to a value that works for you...... or buy new phones :)
 
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