ShoreTel to Cisco SIP trunk

[pndscm]

I have this working dialing from ShoreTel to Cisco phone but not the other way around. I can see the SIP packets in WireShark going from the Cisco system to ShoreTel but the calls fail with a fast busy. I have the Cisco end using UDP.

Any other suggestions?

 

[atascoman]

Is the Cisco using the same SIP port as the Shoretel? I think Shoretel would say you have to have an Ingate Siparator to do a SIP trunk connection. Basically the Shoretel and CIsco would register to the Siparator and it would send the calls to each system.

It also could be a SIP profile setting.

 

[pndscm]

I can call one way so I think the SIP ports are right. I've read some about the Siparator but not sure why it's needed if I can call one way.

 

[atascoman]

It may be that the Shoretel can send calls and the Cisco will accept them, but the Cisco may be sending something in the packet that the Shoretel doesn't recognize. Shoretel even admits their SIP trunking still needs some polish. I think that's why they recommend the Siparator so it does the translation between disparate systems and eliminates any language barrier so to speak.

There is not really a SIP standard yet, so most of the time devices will have different SIP profiles that you have to tweak to get them to register or communicate properly.

 

[nintendo1889]

What do you mean when you say that there isn't really a SIP standard yet?

 

[atascoman]

Basically there is not a set SIP profile/configuration standard that everyone has to use or comply with. Most devices have unique SIP profile configs so you have to tweak your system settings to comply with that device.

When PRI first came out it was the same way, then they came out with the NI2 standard.

 

[nintendo1889]

I understand now.

There's a SIPconnect group and others standards groups trying to make a standard but it doesn't work well imho.

I hooked up an ip pbx to a certain clec's sip trunks which are certified on that pbx, and they are SIPconnect members.

Everything works fine except one little thing, they can't dial their DIDs internally. They're being cheap by pointing our pbx to its own rtp stream for the calls audio (which doesn't work with calls that go out and come back to itself that way. I guess both are to blame, the pbx for not being advanced enough and the provider for being cheap on bandwidth or processing). If using ni2 one could always call their own DIDs with no problem.

The great thing about standards is there are so many to choose from! - Tanenbaum