Remote 1608 no dial tone

[drooney]

I am in the process of relocating one of my offices. Until the IP Office 500 v2 is moved from the old location to the new i want to have my 1608 hand sets connect across a Site2Site VPN and connect to the call manager. What I am seeing is, the phone see's the call manager asks for extension and password but when it registers i see only the extension and feature labels. There is no dial tone at all. Any suggestions?

 

[amriddle01]

You have ports being blocked or an ALG is stopping the RTP traffic, ALWAYS the case with such issues



"No problem monkey socks

 

[holdmusic34]

Turn off enable direct media path. Try that.

Are these ports blocked in a firewall?

>> Port 1719 (H.323 RAS):
Response to a VoIP device registering with IP Office.

>> Port 1720 (H.323/H.245):
Data to a registered VoIP device.

 

[holdmusic34]

Damn Amriddle. Thought i could sneak an answer in before Peter got up!

 

[amriddle01]

You raise a good point though, you do tend to get dialtone even when DMP is the issue, but it still needs turning off either way



"No problem monkey socks

 

[tlpeter]

And again this is a problem with H.323 inspect or blocked ports.
What kind of routers are in between?
Holdmusic, i was painting my henhouse so that is why i was so slow


BAZINGA!

I'm not insane, my mother had me tested!

 

[drooney]

As i stated this is a site to site VPN so we are not blocking any ports. The old site has a sonicwall with the IP 500 located on a VLAN the remote site has a ASA 5510 with the phones sitting on a VLAN. DHCP option 242 is enabled pointing at the call manager at the old site. As a test, I have an ASA at my home office I brought a phone home with me and I am able to connect to the call manager at the old site using site to site VPN.

This one really has me stumped. Any help would be greatly appreciated.

 

[amriddle01]

You can either look into what we have told you and therefor fix the issue or keep asking and it will remain broken Seriosly though this is an issue with RTP traffic being blocked/misdirected, if not directly with rules then indirectly by an ALG, both Sonicwall and Cisco firewalls/routers have them and they cause this issue. The handsets and the system just send to the address they have been told via the gateway they are told, if traffic isn't reaching either device it's the network devices that sit in between you need to look at. It's proven to be the case time after time, at least 10 - 15 times on here alone and it was what we said that proved to be the issue with each and every one



"No problem monkey socks

 

[CptIPO]

on a site to site VPN you are still using routers which see them as separate networks. You must make sure your routing is correct or NAT enabled on WAN of PBX or router. Also Sonicwalls have really powerfull firewalls built in. So make sure you are not blocking ports.

Yes we have VoIP in Cape Town

 

[drooney]

Just spent the last 6 hours on the phone with Cisco... confirming that traffic is NOT being blocked. The Sysmon on the avaya system is showing this in the log for ext 2008 which is the extension i am working with


90638920mS H323Evt: Recv GRQ from 0a030437
90638920mS H323Evt: e_H225_AliasAddress_dialedDigits alias
90638921mS H323Evt: found number <2008>
90638947mS H323Evt: Recv: RegistrationRequest 10.3.4.55; Endpoints registered: 13; Endpoints in registration: 3
90638947mS H323Evt: e_H225_AliasAddress_dialedDigits alias
90638948mS H323Evt: found number <2008>
90638948mS H323Evt: RRQ --- CallSigProtocol is H323AnnexL_P. Go for Avaya 4600IP phone
90638948mS H323Evt: RRQ --- Treat this as a forced login, the phone was probably rebooted (same_cs_addr 0)
90638948mS H323Evt: GK: Unregister endpoint RTS-LI-IPO-01_5162cae155953438 for extension 2008
90638949mS H323Evt: GK: Send URQ
90638950mS H323Evt: RRQ --- Register extn 2008 using product IP_Phone, version 1.2100

 

[drooney]

Still having same issue... no dial tone.. can not dial digits..

 

[tlpeter]

How is this extension configured in the IPO?

BAZINGA!

I'm not insane, my mother had me tested!

 

[drooney]

It is configured as it was when it was at the other site. A user extension.

 

[tlpeter]

Something is blocking it if it worked locally.
So take the phone to the IPO site and connect it.
Show that it works.
Then take it to the remote site and show that it does not work.
Walk away and tell them you do not come back before the network is fixed.


BAZINGA!

I'm not insane, my mother had me tested!

 

[CptIPO]

You should have fixed IP on phone. Also it [ignore] sounds like you have not enabled NAT on WAN of PBX. Although dial tone on the phone comes from the actual phone. So if there is no dial tone but the phone looks like it is working maybe handset is faulty. Dial tone is supplied by phone. Do you get one way speech to trunks as well as internal call or only to internal?

ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town

 

[tlpeter]

No need for a fixed IP address.
The router should take care of the routing (duhh ) and the phone can have an IP address by DHCP locally or by DHCP relay.


BAZINGA!

I'm not insane, my mother had me tested!

 

[hairlessupportmonkey]

The Cisco should do DHCP options too. So , you can set the IPO and port addresses for phones to connect successfully on DHCP.

http://www.ipofficeassistance.com/HOWTO_DHCP_IP_Phones.html

Cisco are natorious for messing with traffic across the VPN. Delete ANY inspect maps for H.323.

If that fails, DELETE the VPN profiles, and re-create them from the VPN Wizard. Reboot both firewalls.

ACSS - SME
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