SOLVED: 7404VNPX outbound local calls engaged

[4f3547]

Hello,

My parents have been unable to call local numbers from their Exetel VOIP service using a Billion 7404VNPX. I closed the related support ticket on 27-07, however I was mistaken because while the issue with mobile calls had been resolved, the issue with local calls remains. Given that this condition was introduced nearly 2 months after their problems started, it's possible that this is an unrelated issue. Here is a summary of the current issue:

Incoming calls to Exetel VOIP number:

• No issues.

Exetel's voice mail service:

• No issues.

Outgoing VOIP calls to mobile numbers:

• No issues.

Outgoing VOIP calls to local numbers:

• Dial tone is OK.
• Dialling is OK.
• Ringback tone is absent, but recipient phone rings normally.
• Call recipient can answer call but cannot hear anything.
• Caller cannot hear anything until after about 50 seconds when a busy signal is heard.
• This occurs using 3 different phones plugged into either registered FXS port separately.
• Call appears in the VOIP modem/router's advanced--> status -->voip call log page, however the start, end and duration fields are blank.
• Call does not appear in the Exetel online VOIP meter.

The procedures I have tried so far to solve this problem are:

• Re-flashed VOIP modem/router firmware using serial cable and performed a factory reset (using Billion Recovery Software).
• Updated firmware using the VOIP modem/router's web interface to versions 5.53.s6.b1, 6.02a, 6.02b and 6.02c and reset to factory defaults multiple times.
• Changed codec preferences of VOIP modem/router to use either G.729, PCMU(G.711 A-Law) or PCMU(G.711 u-Law) exclusively.
• Swapped out all phone cables.
• Tested with 3 phones (previously working) using both of the router's FXS registered ports.
• Tested SIP client (X-Lite softphone) using Exetel VOIP account which worked for all types of calls with no issues.
• Tested the Exetel VOIP service using Thomson SpeedTouch 780WL VOIP modem/router which worked for all types of calls with no issues.
• Tested Billion 7404VNPX using a different VOIP provider (FaktorTel) which worked for all types of calls with no issues.
• Contacted Billion's Online Helpdesk and followed all suggested troubleshooting steps.

I was looking through the router's event logs earlier and I noticed a few differences in the logs recorded during successful calls made to mobiles compared with the failed calls made to local numbers. It appears that during some of the unsuccessful INVITE messages, an invalid URI is provided in a "From" field. I apologise if this terminology is incorrect! I don't know where this request/response originates from or what it means, but I will post some excerpts from the logs to demonstrate. Thank you!

Excerpt from successful outbound call to mobile:

Code: Select all

SIP Event EvProceeding in State StateCalling for 03XXXXXXXX (0) on Dialog 1378230
SIP state now StateCalling on FXS 0
sip_SipSelectThread_?: [SipTransportLayer::messageReceived]-> Entered this function
sip_SipSelectThread_?: SIP/2.0 183 Session Progress

Call-ID: 1210809dd@220.233.XXX.XXX

Contact: <sip:611831417XXXXXX@58.96.1.1>

Content-Length: 218

Content-Type: application/sdp

CSeq: 235 INVITE

From: "03XXXXXXXX"<sip:03XXXXXXXX@58.96.1.2>;tag=171520e8h

Record-Route: <sip:58.96.1.2;lr=on;ftag=171520e8h>

To: <sip:0417XXXXXX@58.96.1.2>;tag=3a600101-18d5dc

User-Agent: Quintum/1.0.0

Via: SIP/2.0/UDP 220.233.XXX.XXX:5060;received=220.233.XXX.XXX;branch=z9hG4bK178570et1;rport=5060

Quintum: 07120107010000af8306001e038084881e028488



v=0

o=Quintum 61782 12530 IN IP4 58.96.1.1

s=VoipCall

c=IN IP4 58.96.1.1

t=0 0

m=audio 11936 RTP/AVP 8 101

c=IN IP4 58.96.1.1

a=rtpmap:8 pcma/8000/1

a=ptime:20

a=rtpmap:101 telephone-event/8000/1

a=sendrecv


sip_SipSelectThread_?: 

Excerpt from unsuccessful outbound call to local number:

Code: Select all

SIP Event EvProceeding in State StateCalling for 03XXXXXXXX (0) on Dialog 136ece0
SIP state now StateCalling on FXS 0
Num_retries in choose_server entry = 0
sip_SipSelectThread_?: [SipTransportLayer::messageReceived]-> Entered this function
sip_SipSelectThread_?: SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 220.233.XXX.XXX:5060;received=220.233.XXX.XXX;branch=z9hG4bK138080bol;rport=5060

From: <sip:@58.96.1.2>;tag=177530eq5

To: <sip:XXXXXXXX@58.96.1.2>;tag=as67791130

Call-ID: 1121208oh@220.233.XXX.XXX

CSeq: 172 INVITE

User-Agent: ExetelVoip

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

Contact: <sip:03XXXXXXXX@58.96.1.13>

Content-Type: application/sdp

Content-Length: 281



v=0

o=root 3447 3447 IN IP4 58.96.1.13

s=session

c=IN IP4 58.96.1.13

t=0 0

m=audio 16314 RTP/AVP 8 0 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


sip_SipSelectThread_?: 

Thank you very much for your assistance!

[MCCMikey]

Exetel does not support Billion routers. You'll probably need to find another VoIP service provider, or buy a NetComm product.

[Dazzled]

Grr, I just made a very long reply, and the @!$% timeout function swallowed it.

Anyway, it's not so much that Exetel doesn't support Billion, as it's Billion having some router firewall eccentricities, particularly with DMZ and ALG.

The handshake rigmarole with registered proxy servers is needed to get around your router firewall dropping uninvited outside connections - essential to keep the black hats out.

The problem here is a firewall problem between the router and built in ATA (the INVITEs are being screwed up). The firewall is correctly passing signals to the softphone on the network, and it is really only another ATA. I would start by finding the Billion ALG setting and reversing it.

There is a 7404 help sheet at http://www.exetel.com.au/pdf/Billion7404setup.pdf

[jokiin]

[*]Changed codec preferences of VOIP modem/router to use either G.729, PCMU(G.711 A-Law) or PCMU(G.711 u-Law) exclusively.

set your preference by all means but don't set any codec for exclusive use as some calls will fail and others will work, put them in order of preference but make them all available

[sable]

There is a 7404 help sheet at http://www.exetel.com.au/pdf/Billion7404setup.pdf

Dazzled, this is the old 7404 the OP is using the model with the new interface, the SIP ALG setting can be found at Advanced/Configuration/Firewall/General Settings

Mine is set to "Enable" and have no issues, however Firewall is set to Disable.

I also note in the 7404 help sheet above the Expire Time is set to the old 240 seconds the latest recommendation after the changes Exetel made to the sip server is 1800 seconds.

[jokiin]

There is a 7404 help sheet at http://www.exetel.com.au/pdf/Billion7404setup.pdf

Dazzled, this is the old 7404 the OP is using the model with the new interface, the SIP ALG setting can be found at Advanced/Configuration/Firewall/General Settings

Mine is set to "Enable" and have no issues, however Firewall is set to Disable.

I also note in the 7404 help sheet above the Expire Time is set to the old 240 seconds the latest recommendation after the changes Exetel made to the sip server is 1800 seconds.

240 is still fine to use and wouldn't be related to the problem at hand, things do change from time to time though so worthwhile double checking settings anyway as some of the help guides around are probably out of date

[Dazzled]

Sable thanks for reminding me about the new interface; I had forgotten that when I referred to the old helpsheet.

As the OP's softphone works, something is amiss between the router iptables and the on-board Billion ATA application. I still suspect ALG rather than a particular codec or registration time. Setting Firewall to Disable may bypass that; certainly try it.

[sable]

240 is still fine to use and wouldn't be related to the problem at hand, things do change from time to time though so worthwhile double checking settings anyway as some of the help guides around are probably out of date

Jokiin

I was not offering the expire time setting as a solution to the OP's issue, just pointing out that the current recommenation for those experiencing the problem per passanm is 1800 see

viewtopic.php?f=58&t=34160

Since the sip server was changed causing the problems with Billions and VoIP my experience has been that some accounts are not affected and work fine with 240, others are still having problems even today with 1800 and others are now ok after changing to 1800. JI ust cannot understand that the problem is not consistent!

[jokiin]

I was not offering the expire time setting as a solution to the OP's issue,

yeah knew where you were coming from, even if I wasn't clear in my reply

Just cannot understand that the problem is not consistent!

welcome to Billion, not my favorite devices

[jokiin]

As the OP's softphone works, something is amiss between the router iptables and the on-board Billion ATA application. I still suspect ALG rather than a particular codec or registration time. Setting Firewall to Disable may bypass that; certainly try it.

Neither the firewall or the SIP ALG settings should have any impact on the internal SIP client, it is a Billion though so hard to say never, I would still expect it to be something else that is tripping it up though

[zencloud]

I've been recently having the same problem with a 7404VGPX.

[pasanm]

Hello,

My parents have been unable to call local numbers from their Exetel VOIP service using a Billion 7404VNPX. I closed the related support ticket on 27-07, however I was mistaken because while the issue with mobile calls had been resolved, the issue with local calls remains. Given that this condition was introduced nearly 2 months after their problems started, it's possible that this is an unrelated issue. Here is a summary of the current issue:

Incoming calls to Exetel VOIP number:

• No issues.

Exetel's voice mail service:

• No issues.

Outgoing VOIP calls to mobile numbers:

• No issues.

Outgoing VOIP calls to local numbers:

• Dial tone is OK.
• Dialling is OK.
• Ringback tone is absent, but recipient phone rings normally.
• Call recipient can answer call but cannot hear anything.
• Caller cannot hear anything until after about 50 seconds when a busy signal is heard.
• This occurs using 3 different phones plugged into either registered FXS port separately.
• Call appears in the VOIP modem/router's advanced--> status -->voip call log page, however the start, end and duration fields are blank.
• Call does not appear in the Exetel online VOIP meter.

The procedures I have tried so far to solve this problem are:

• Re-flashed VOIP modem/router firmware using serial cable and performed a factory reset (using Billion Recovery Software).
• Updated firmware using the VOIP modem/router's web interface to versions 5.53.s6.b1, 6.02a, 6.02b and 6.02c and reset to factory defaults multiple times.
• Changed codec preferences of VOIP modem/router to use either G.729, PCMU(G.711 A-Law) or PCMU(G.711 u-Law) exclusively.
• Swapped out all phone cables.
• Tested with 3 phones (previously working) using both of the router's FXS registered ports.
• Tested SIP client (X-Lite softphone) using Exetel VOIP account which worked for all types of calls with no issues.
• Tested the Exetel VOIP service using Thomson SpeedTouch 780WL VOIP modem/router which worked for all types of calls with no issues.
• Tested Billion 7404VNPX using a different VOIP provider (FaktorTel) which worked for all types of calls with no issues.
• Contacted Billion's Online Helpdesk and followed all suggested troubleshooting steps.

I was looking through the router's event logs earlier and I noticed a few differences in the logs recorded during successful calls made to mobiles compared with the failed calls made to local numbers. It appears that during some of the unsuccessful INVITE messages, an invalid URI is provided in a "From" field. I apologise if this terminology is incorrect! I don't know where this request/response originates from or what it means, but I will post some excerpts from the logs to demonstrate. Thank you!

Excerpt from successful outbound call to mobile:

Code: Select all

SIP Event EvProceeding in State StateCalling for 03XXXXXXXX (0) on Dialog 1378230
SIP state now StateCalling on FXS 0
sip_SipSelectThread_?: [SipTransportLayer::messageReceived]-> Entered this function
sip_SipSelectThread_?: SIP/2.0 183 Session Progress

Call-ID: 1210809dd@220.233.XXX.XXX

Contact: <sip:611831417XXXXXX@58.96.1.1>

Content-Length: 218

Content-Type: application/sdp

CSeq: 235 INVITE

From: "03XXXXXXXX"<sip:03XXXXXXXX@58.96.1.2>;tag=171520e8h

Record-Route: <sip:58.96.1.2;lr=on;ftag=171520e8h>

To: <sip:0417XXXXXX@58.96.1.2>;tag=3a600101-18d5dc

User-Agent: Quintum/1.0.0

Via: SIP/2.0/UDP 220.233.XXX.XXX:5060;received=220.233.XXX.XXX;branch=z9hG4bK178570et1;rport=5060

Quintum: 07120107010000af8306001e038084881e028488



v=0

o=Quintum 61782 12530 IN IP4 58.96.1.1

s=VoipCall

c=IN IP4 58.96.1.1

t=0 0

m=audio 11936 RTP/AVP 8 101

c=IN IP4 58.96.1.1

a=rtpmap:8 pcma/8000/1

a=ptime:20

a=rtpmap:101 telephone-event/8000/1

a=sendrecv


sip_SipSelectThread_?: 

Excerpt from unsuccessful outbound call to local number:

Code: Select all

SIP Event EvProceeding in State StateCalling for 03XXXXXXXX (0) on Dialog 136ece0
SIP state now StateCalling on FXS 0
Num_retries in choose_server entry = 0
sip_SipSelectThread_?: [SipTransportLayer::messageReceived]-> Entered this function
sip_SipSelectThread_?: SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 220.233.XXX.XXX:5060;received=220.233.XXX.XXX;branch=z9hG4bK138080bol;rport=5060

From: <sip:@58.96.1.2>;tag=177530eq5

To: <sip:XXXXXXXX@58.96.1.2>;tag=as67791130

Call-ID: 1121208oh@220.233.XXX.XXX

CSeq: 172 INVITE

User-Agent: ExetelVoip

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

Contact: <sip:03XXXXXXXX@58.96.1.13>

Content-Type: application/sdp

Content-Length: 281



v=0

o=root 3447 3447 IN IP4 58.96.1.13

s=session

c=IN IP4 58.96.1.13

t=0 0

m=audio 16314 RTP/AVP 8 0 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


sip_SipSelectThread_?: 

Thank you very much for your assistance!

Hi

you have a ported number.

In your members facility allow callerid to be display. Currently you have disable callerid display.

There's a problem with upstream carrier not accepting calls without callerid.

As a troublshooting setup

enable callerid from members facility and check whether any call success without issues.

Thanks

[jokiin]

Hi

you have a ported number.

In your members facility allow callerid to be display. Currently you have disable callerid display.

There's a problem with upstream carrier not accepting calls without callerid.

As a troublshooting setup

enable callerid from members facility and check whether any call success without issues.

Thanks

makes sense now that none of the other suggestions were getting anywhere, doesn't make much sense that you can't block caller ID though, pass it sure but you should still be able to not show it selectively, is this being addressed with the carrier as I can see that it would be a problem for some people

[pasanm]

Hi

you have a ported number.

In your members facility allow callerid to be display. Currently you have disable callerid display.

There's a problem with upstream carrier not accepting calls without callerid.

As a troublshooting setup

enable callerid from members facility and check whether any call success without issues.

Thanks

makes sense now that none of the other suggestions were getting anywhere, doesn't make much sense that you can't block caller ID though, pass it sure but you should still be able to not show it selectively, is this being addressed with the carrier as I can see that it would be a problem for some people

Hi

I'm currently working with our upstream carrier regarding this.

[sable]

Changed my setting to disable caller id.

Have not had a landline to call yet, but works fine on Exetel to Exetel did.