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GXW-4108 problem

Warloch
Posts: 20
 Member Since:
2007-04-08
Submitted by Warloch on Mon, 02/16/2009 - 4:31pm.

I've got a Trixbox system that we has been using a GXW-4108 successfully for months. Originally we had 5 lines coming in and I had it setup as a trunk using method 2 in the quick start guide so that all of the calls were sent to the same caller ID. We recently added 2 more lines for a separate company in the same building. I changed the UserID to ch1-5:NXXNXX9850;ch6-7:NXXNXX9860. So far incoming calls are routed properly. I also added a second trunk. I used the following peer details:

port=5070
context=from-trunk
host=192.168.8.9
type=peer
insecure=very
qualify=yes
dtmfmode=auto

I used port 5070 since that is the port number for the 6th channel. I also changed the Port Scheduling Schema for Round Robin to rr:1-5;rr:6-7;rr:8; since we want to use the first 5 ports for company A and port 6 & 7 for company B. This will use the caller ID from our provider. I setup outbound routing so that if they dialed a straight 10 digit call (10-digit dialing is mandatory here) it would go out the trunk for company A. If they prefix the call with an 8 it goes out the trunk for company B. So far outbound routing appears to work properly.

Now for the problem. The system seems to work fine for several hours and then it's like the gateway just stops responding. Calls can be made in but not out. If I reboot the gateway it starts working again. When it quits I can do a sip show peers in the console and the trunks show as normal. I can do a sip show channels and no channels are shown. I can log into the gateway so it does not appear to be locked up. Even when it doesn't process calls, the debug shows it responding to the SIP messages. To test it, since I'm not on site, I setup an inbound route that forwards to an external number. I do have debug traces for a both a working and a non-working call. In both cases the incoming call works fine. Can anyone help me?

John

‹ Dead GXP2000 tftp not working GXP-2010's lock up randomly. ›

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ragoley
Posts: 3
 Member Since:
2008-05-07
Same problem here
Tue, 04/21/2009 - 9:42am

I am having the same trouble but I am not using the second sip server. I am sharing all lines with the same settings you have above. It worked fine for a while. I initially had to reboot it a little. Now I have perfect incoming but all outbound calls fail. The number is called and the callee can answer it. There is no audio on either side of the call. The person calling does not know the call was completed at all. The person receiving the call just gets silence. I did logs and wireshark captures for support. I have not heard back on those yet. I am now using 1.0.1.24 firmware that was given to me by support. It has not helped this issue. It just changed it a little... Does anyone have a GXW-4108 working correctly for inbound and outbound calls without issues like this? If so, I would really like to know what your basic config is.

Robert


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Gerrymad
Posts: 111
 Member Since:
2007-11-21
Same issue here. This device
Tue, 04/21/2009 - 10:28am

Same issue here. This device just doesn't seem to work that well for this purpose. About once a week I have to reboot it to get it to work. From the trixbox side everything appears normal. Logging into the GXW also shows everything normal. Outbound calls just don't go through.

It is just in a test environment (I am glad it is in that location), but it is still bothersome. Ultimately I think it is off to Rhino to get a proper card.


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madni.abbasi
Posts: 2
 Member Since:
2009-05-06
Can't receive Call In
Wed, 05/27/2009 - 4:28am

I am using Gramstream GXW-4108. I have one PSTN line. According to my current setting, I am able to select PSTN line using my extension to call out (Outbound calls). It's working successfully but when I call to my PSTN line using another PSTN line I dont get any call on my IP extension.

Environment:

Trixbox CE 2.6 (CentOS release 5.2 (Final))
Zoiper Softphone
GXP2000
GXW-4108

Please advice


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friesen
Posts: 22
 Member Since:
2007-09-04
GXW4108 stops making or taking calls daily
Tue, 07/14/2009 - 10:07am

I'm having the same problem, it seems to have gotten worse every time i have upgraded the firmware. i have tried the current firmware (1.0.1.25) and the beta firmware (1.2.1.5) and it seems for some reason my GXW4108 stops accepting calls and will not allow calls to be made after about a day or two of use. rebooting it will resolve the problem. it's getting very frustrating. can someone please help.


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joshpatten
Posts: 733
 Member Since:
2007-01-20
friesen wrote: it seems to
Tue, 07/14/2009 - 6:53pm
friesen wrote:
it seems to have gotten worse every time i have upgraded the firmware

Welcome to Grandstream. If you plan on sticking with them prepare for it to get consistently worse. The developers have an extreme case of "Feature Creep" that tends to break a lot of things along the way. My advice: get an Audiocodes or one of the PCI cards.


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Techmann
Posts: 1
 Member Since:
2009-07-28
GXW4108 stops making or taking calls daily
Tue, 07/28/2009 - 5:46pm

Running 1.0.1.25 firmware. Unit is hanging several times a day. On Asterisk CLI when trying to do an outbound call, I can see that Asterisk is sending a a normal connect msg to GWX but it doesn't responsd back. Doing a "sip show peers" shows the GXW to still be regiestered with Asterisk. When the unit hangs, all Syslog debug messages stop being sent from unit almost as if the Ethernet port (LAN) has stopped responding. However, I am still able to log into unit and reboot it so Ethernet port must still be working somewhat. As soon as I reboot the unit, everything starts working normally. Does anybody have a fix for this and what else should I try? Please help.


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Feb
Posts: 1
 Member Since:
2009-03-14
Sorry my English I had such
Fri, 07/31/2009 - 4:48am

Sorry my English

I had such problem described above, but after firmware, the problem while has faded.
Software Version: Program - 1.0.1.25 Loader - 1.1.3.4 Boot - 1.1.3.2.

As has noted that FreePBX somehow not correctly sampled the outbound router. That is, in rules of a set one has been pointed router, and he sampled another. I have swapped entering PSTN line on gateway and swapped accounts on it. But the following problem is watched now.
I show the my log:


INVITE sip:7777@192.168.XXX.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.XXX.23:5060;branch=z9hG4bK5373a9f0e84341a7
From: "unknown";tag=52553eb1d8b41cc4
To:
Contact:
Supported: replaces, timer, path
Call-ID: 3da2ffb6fd11cd808ac5cfc655f47384@192.168.XXX.23
CSeq: 15259 INVITE
User-Agent: Grandstream GXW4108 (HW 1.0, Ch:6) 1.0.1.25
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 313

v=0
o=41380 8006 8000 IN IP4 192.168.XXX.23
s=SIP Call
c=IN IP4 192.168.XXX.23
t=0 0
m=audio 5028 RTP/AVP 0 8 4 18 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11


--- (13 headers 15 lines) ---
Sending to 192.168.XXX.23 : 5060 (no NAT)
Using INVITE request as basis request - 3da2ffb6fd11cd808ac5cfc655f47384@192.168.XXX.23
Found peer 'GXW4108p2-41370'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.XXX.23:5028
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xfae (gsm|ulaw|alaw|g726|adpcm|lpc10|g729|speex|ilbc), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - (ulaw|alaw|g729|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.XXX.23:5028
Looking for 7777 in from-trunk-custom (domain 192.168.XXX.1)
list_route: hop:


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.XXX.23:5060;branch=z9hG4bK5373a9f0e84341a7;received=192.168.XXX.23
From: "unknown";tag=52553eb1d8b41cc4
To:
Call-ID: 3da2ffb6fd11cd808ac5cfc655f47384@192.168.XXX.23
CSeq: 15259 INVITE
User-Agent: Roks PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

From my log it is visible that account (v=o; o=41380) one and peer another (Found peer ' GXW4108p2-41370 ')
Why so?
Please help.
Feb


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