NEW SERVER INFORMATION IN ALL SETUP GUIDES

Please note that currently many of the following setup guides show the Sip Server Info as voipuser.org As of August 8th 2005 this has been changed due to a change of server.

Please change the SIP proxy domain (sometimes called SIP domain, or SIP registrar) from:-

voipuser.org

to

sip.voipuser.org

We will try and change them all eventually but in the meantime please make the substitution yourself

Hi

As a new user, having just spent a happy few days fixing an asterisk "retries" problem by reverting to voipuser.org, I'd like to see someone's working sip.voipuser.org asterisk config before I make the change!

Any help or a fix to the asterisk setup guide gladly welcomed but I just can't get sip.voipuser.org to work myself.

Do we have to change the hostid as well as the domain, or just one of them?

Many thanks
Jon

I am getting some strange results too. I am testing my setup for the first time with Voipuser. I have both X-Lite and Cisco 7940 with 4.4 firmware. The Cisco cannot make outgoing PSTN calls with the new proxy sip.voipuser.org, as it times out on the invite message. The old proxy voipuser.org works OK. However the X-Lite seems to work with either proxy. Both the Cisco and X-Lite can successfully connect to FWD echo test using either proxy. Can anyone shed any light on this? Could it be the firmware on the Cisco, as I still use the version 4.4, the last of the unprotected versions? I am a bit wary of upgrading because of possible problems.

Thank you,

alistair

Yes, I understand the concerns, partuclarly where some of you have spent quite a number of "happy hours" configuring to get a working system!

First point - don't worry, we won't switch anything over until we're 100% satisfied that it's working OK.

So if you do have problems, please switch back to the old URL of voipuser.org but please let us know so that we can look into it.

If it's a relatively easy thing to do, it would be really helpful if you could at least try changing the server URL in your configuration. If it doesn't work, you can always change it back. But again, please let us know so that we can look into it.

Dean

Dean,

Thanks for the reply. I shall try a few ethereal traces to find out what is happening at my end and let you know the outcome.

Just successfully made a call to family in UK and got a call back via my 0844 number. This was on the Cisco with the old proxy address. Hopefully we can find out what is the problem with the new proxy.

alistair

As promised I have investigated this further and here are my findings, a bit long I'm afraid. I didn't actually complete the PSTN calls, just let the phone ring to avoid costs.

I have tested X-Lite and Cisco 7940 using both sip.voipuser.org and voipuser.org proxies. The behavior is different depending on the type of call - outbound to PSTN or outbound to a SIP number, FWD echo test in this case.

On X-lite both types of call are successful with either proxy
On Cisco both types of call are successful with voipuser.org proxy. With sip.voipuser.org proxy only SIP calls are successful.

The basic difference in behavior is that X-Lite send invites to ***@voipuser.org using either proxy, but Cisco sends invites to ***@voipuser.org or ***@sip.voipuser.org depending on what SIP proxy is configured.

Calling SIP numbers

With X-Lite using either proxy when calling a SIP number the sequence is:

Request: INVITE sip:**393613@voipuser.org,
Status: 100 trying -- your call is important to us

Result call successful

With Cisco using voipuser.org proxy the sequence is the same

Result call successful

With Cisco using sip.voipuser.org proxy the sequence is:

Request: INVITE sip:**393613@sip.voipuser.org,
Status: 100 trying -- your call is important to us

Result call successful

Calling PSTN numbers

With X-lite using either proxy when calling a PSTN number the sequence is:

Request: INVITE sip:003496565****@voipuser.org
Status: 407 Proxy Authentication Required
Request: ACK sip:003496565****@voipuser.org
Request: INVITE sip:003496565****@voipuser.org
Status: 100 trying -- your call is important to us

Result call successful

With Cisco using voipuser.org when calling PSTN number the sequence is:

Request: INVITE sip:003496565****@voipuser.org
Status: 407 Proxy Authentication Required
Request: ACK sip:003496565****@voipuser.org
Request: INVITE sip:003496565****@voipuser.org
Status: 100 trying -- your call is important to us

Result call successful

With Cisco using sip.voipuser.org when calling PSTN number the sequence is:

Request: INVITE sip:003496565****@sip.voipuser.org
Request: INVITE sip:003496565****@sip.voipuser.org
Request: INVITE sip:003496565****@sip.voipuser.org
Request: INVITE sip:003496565****@sip.voipuser.org
etc till timeout

Result call fails

I hope this sheds some light on the situation. One of my clients may be non standard, probably the Cisco phone (old firmware?).

Perhaps the SIP experts could comment.

alistair

Thanks Alistair - just to let you know I've seen this, just not had enough time to digest and look into it. Will do so over the next day or so and come back to you.

Like I say, don't worry, we won't switch servers on you until we're happy everything is A OK.

Dean

Not directly to your subject but may give you an idea/pointer.
~~~
Some hard-phones seem to have most curious configuration interfaces (presumably keeping code small).

For example, quite a few of the Grandstream range abuses the Outgoing Proxy field to allow for the kind of setup Voipuser is moving towards. Maybe the CISCO has something similar.

eg, to get to Voipuser with the new 'sip.' proxy style on a Grandstream, I must set:
~~~
sip server: voipuser.org
outbound proxy: sip.voipuser.org
~~~~
Using Grandstream FAQ from http://www.grandstream.com/y-faq.htm

3. What if my SIP URI domain is different from the SIP proxy server FQDN (Fully Qualified Domain Name)?
With firmware 1.0.3.60 and later, you can put the your SIP URI domain name into the SIP Server field, and put the actual sip server FQDN into Outbound Proxy field. The phone will use the domain name in SIP Server as part of SIP URI but send and receive SIP messages through the SIP
proxy server defined in the Outbound Proxy field.
~~~~
Elsewhere they explicitly say to leave the Outbound Proxy field blank if using STUN. I don't know what you are meant to do if you need a real Outbound Proxy and the domain and SIP FQND are different?

Stephen

Quote:

sip server: voipuser.org outbound proxy: sip.voipuser.org

That is bizarre, but I suspect a result of bad use of terminology on Grandstream's part more than anything else?

Quote:

That is bizarre, but I suspect a result of bad use of terminology on Grandstream's part more than anything else?

It is bad use of terminology, but I don't think that is the root of it. The original phone interface presumably did not allow at all for a difference between sip server and realm. When they needed to, constraints prevented them doing a 'sensible' job and adding an extra field. Maybe the contraints were available memory, or ..., or just not enough good programmers to hand that understood what they were doing.

At least some of these constraints are likely to apply to other limited resource hard-phones -- and it is possible that they took similar (probably not identical) shortcuts to in their solution. (Also possible they just didn't provide the feature at all even in curious form, just hope you'll buy a newer version of their phone.)

Stephen

Stephen,

Thank you for your input. I have tried what you suggested and calls to PSTN are successful. However I am not sure if the outbound proxy setting sip.voipuser.org is actually doing anything apart from converting my 2 line phone to a single line. (Only one outbound proxy is allowed, and FWD doesn't like it) It registers with voipuser.org and sends invites there. No sign of sip.voipuser.org in the trace apart from some unsuccessful DNS lookups.

I remember having similar problems with Babble. Their domain was babble.net and the SIP proxy sip.bon.net. They fixed this at the server end and told me that the Cisco SIP implementation was not standard.

P.S. I have just been reading the Cisco setup manuals and they say input the SIP proxy IP address in numerical form. This of course works perfectly at the moment as voipuser.org and sip.voipuser.org point to the same IP address 216.127.66.119. Perhaps this is the simplest solution.

alistair

I am now testing with a Linksys PAP2 and have to say that the results are the same as with the Cisco, i.e. calls to PSTN fail with sip.voipuser.org as the SIP proxy.

alistair

Quote:

I am now testing with a Linksys PAP2 and have to say that the results are the same as with the Cisco, i.e. calls to PSTN fail with sip.voipuser.org as the SIP proxy.

same for sipura 1001

Had some issues with voipbuster so tried back the voipuser and had proxy issues:

Outgoing calls to PSTN works with voipuser.org on Xlite and spa2000 and failing with sip.voipuser.org as the SIP proxy.

May i ask you all if the failed calls are only failing to PSTN or also failing to other VoIP users ?

Tj