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vitelity trunk


nathans
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Hi,

 

I'm having a tough time getting a Vitelity trunk to work. It used to work but something happen and it no longer receives calls or makes them. Vitelity has checked it thousands of time but no luck. Tried both Sip registration and getaway. Now i'm in the point where calls out ring the phone but with only 1 way audio (snom to remote phone)

Incoming calls do nothing. This is the log of an incoming one:

 

5] 2012/02/11 01:05:28: Identify trunk (IP address/port and domain match) 6

[6] 2012/02/11 01:05:28: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 INVITE

Content-Length: 0

 

[6] 2012/02/11 01:05:36: SIP Rx udp:64.2.142.15:5060:

CANCEL sip:36144423xx@192.168.88.127:5060 SIP/2.0

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport

From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 

[6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

Content-Length: 0

 

[5] 2012/02/11 01:05:36: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com

[6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

Contact: <sip:nsandler@75.149.181.125:5060;transport=udp>

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

[3] 2012/02/11 01:05:36: Via and source address are empty for SIP/2.0, cannot send reply

 

The call should route to extension 70 (as it is correctly trying to) but nothing happens. Any ideas what "Via and source address are empty for SIP/2.0, cannot send reply" means?

 

Thanks

Nathan

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Hi,

 

I'm having a tough time getting a Vitelity trunk to work. It used to work but something happen and it no longer receives calls or makes them. Vitelity has checked it thousands of time but no luck. Tried both Sip registration and getaway. Now i'm in the point where calls out ring the phone but with only 1 way audio (snom to remote phone)

Incoming calls do nothing. This is the log of an incoming one:

 

5] 2012/02/11 01:05:28: Identify trunk (IP address/port and domain match) 6

[6] 2012/02/11 01:05:28: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 INVITE

Content-Length: 0

 

[6] 2012/02/11 01:05:36: SIP Rx udp:64.2.142.15:5060:

CANCEL sip:36144423xx@192.168.88.127:5060 SIP/2.0

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport

From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

 

[6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

Content-Length: 0

 

[5] 2012/02/11 01:05:36: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com

[6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060

From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510

To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a

Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15

CSeq: 102 CANCEL

Contact: <sip:nsandler@75.149.181.125:5060;transport=udp>

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

[3] 2012/02/11 01:05:36: Via and source address are empty for SIP/2.0, cannot send reply

 

The call should route to extension 70 (as it is correctly trying to) but nothing happens. Any ideas what "Via and source address are empty for SIP/2.0, cannot send reply" means?

 

Thanks

Nathan

 

We have many vitelity trunks without a problem.

 

Sounds to me like a NAT/FIREWALL issue. make sure you have all ports forwarded and if you are using nat, you may need to adjust the routting in admin/ports.

Also make sure you don't have a sip alg in the way.

BTW what firewall/router are you using?

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Thanks Moishe for the answer.

 

The snomone is indeed behind a firewall but it has all of the ports forwarded to it (kind of DMZ)

We are using a Peplink router. For the NAT issue, we have the local/public IP in the port part of the configuration.

Playing around with some settings in the trunk I managed to make outgoing calls. The incoming ones are still failing. Form the log it seems like the snomone is getting and recognizing the incoming call, even sending it to the right AA account but nothing happens:

 

 

[5] 2012/02/12 22:39:19: Domain trunk vitelity@sxxxx.beta-brain.com sends call to 70 in domain snom.beta-brain.com

[6] 2012/02/12 22:39:19: SIP Tx udp:64.2.142.xxx:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK3f21e420;rport=5060

From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.xx>;tag=as44e70cd8

To: <sip:361xxxxxxx@75.149.181.xxx:5060>;tag=288c11fec4

Call-ID: 50499c6c0b573e8b62cdae0201ab3d25@64.2.142.15

CSeq: 102 CANCEL

Contact: <sip:nsandler@75.149.181.xxx:5060;transport=udp>

User-Agent: snom-PBX/2011-4.3.0.5021

Content-Length: 0

 

[3] 2012/02/12 22:39:19: Via and source address are empty for SIP/2.0, cannot send reply

[1] 2012/02/12 22:39:21: Timeout: Call 44 not found

 

 

Any chance you can share with me your trunk definitions for any a Vitelity line?

Also the LAG your are referring is in the router or in the snomone?

 

Thanks a lot!!

Nathan

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That would also concern me. Maybe there is a UDP fragmentation problem? If you have Wireshark on the PBX host, it would be easy to see if that is the case.

See here http://www.peplink.com/index.php?view=faq&id=124&path=16 how to turn off sip alg in your router.

 

With most routers I find a need to use the IP routing list in admin, settings, ports, sip. We try to avoid NAT whenever possible.

 

also port forward 5060/5061 tcp and udp, your SnomOne's rtp port range udp, and vitelity's rtp port range udp. I beleive their default is 10000-20000.

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  • 5 months later...

After a few months, we have yet to make this work. I'm seeing something strange on the logs that maybe a clue to somebody out there. Its seems like the incoming call is detected and recognized and it fails to be tranfer to the extension 70 which is the incoming send all extension. Have no clue why....the calling party juts hears silence for about 10 seconds and then gets a fast busy.

Any ideas anyone? Thanks!!

 

[8] 2012/08/08 11:58:02: Trunk: Check if the call to +1361xxxxxx comes from the cell phone +1305xxxxxxx

[8] 2012/08/08 11:58:02: To is <sip:+1361xxxxxxx@75.149.181.121:5060;user=phone>, user 0, domain 1

[8] 2012/08/08 11:58:02: Send call to extension ERE returned 70

[5] 2012/08/08 11:58:02: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com

[5] 2012/08/08 11:58:04: SIP Tx udp:64.2.142.15:5060:

INVITE sip:305xxxxxxx2@64.2.142.15 SIP/2.0

Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;rport

From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa

To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc

Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15

CSeq: 30655 INVITE

Max-Forwards: 70

Contact: <sip:nsandler@75.149.181.121:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Type: application/sdp

Content-Length: 314

 

v=0

o=- 1472081629 1472081629 IN IP4 75.149.181.121

s=-

c=IN IP4 75.149.181.121

t=0 0

m=audio 60702 RTP/AVP 18 0 8 101

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2012/08/08 11:58:04: SIP Rx udp:64.2.142.15:5060:

SIP/2.0 491 Request Pending

Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;received=75.149.181.121;rport=5060

From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa

To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc

Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15

CSeq: 30655 INVITE

User-Agent: packetrino

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

Supported: replaces

Content-Length: 0

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[5] 2012/08/08 11:58:02: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com

[5] 2012/08/08 11:58:04: SIP Tx udp:64.2.142.15:5060:

INVITE sip:305xxxxxxx2@64.2.142.15 SIP/2.0

 

What is 305xxxxxxx2@64.2.142.15? Is this an internal number or something else? If it is an internal number then either it should ring or answer (if it is some AA/ACD/Hunt etc).

 

The response is sent by "User-Agent: packetrino". Does this ring any bell?

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Ok, not very clear about the call flow here. This log that you attached is the outbound INVITE from PBX. Can you check if some call forwarding (or service flag) is set on the account 70. Maybe the call is being forwarded to somewhere.

 

In any case, please enable complete logging (at level 9 for pretty much everything & SIP call logging, no need of REGISTER, SUB/NOT) and PM the log to me.

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  • 3 weeks later...
  • 7 months later...

For future Vitelity trunk customers you will have to add the IP address of sipXX.vitelity.net and outbound.vitelity.net in the "Explicitly list addresses for inbound traffic:" also set Secure: to yes , for inbound calls to work.

 

To find the IP addr of these DNS do a ping on the cmd line.

 

Does the "Explicitly list addresses for inbound traffic:" setting accept IP address blocks; b/c, according to Vitelity, Media & SIP signalling for calls may originate from within the following IP ranges:

64.2.142.0/24

66.241.96.0/24

66.241.99.0/24

66.241.111.0/24

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