06.10.2006 21:08

Forum: Miscellaneous

I think that means we are the only ones that are commenting on it, which is not the same as the only ones experiencing it!

07.08.2006 07:04

Forum: Miscellaneous

I think we both need this function, however untill Pascal can make time to get through some of these issues, we may may have to suffer. I suspect there may be some config tweaks as a work around.

Caller ID 23.03.2006 06:04

Forum: Providers

I have tested this feature now and I would estimate that the caller ID informatio will probably not be passed by many SIP providers.

I can however set you up with an account that we can give you any caller ID you want, however I do not think anybody will guarantee Caller ID to all destinations. If you are talking about caller ID to USA. it will appear on more than 95% of calls from my experience. No name however, just number.

Mark
Supernettel.com

14.03.2006 18:52

Forum: Terminal Equipment

This error message has now changed to a 403 error! I can log in to an accoiunt with a Sipura, but not with x-pro. Funny thing about this is that it registered fine a couple of days ago. I have tried FQDN and IP address.

404 errors 14.03.2006 06:21

Forum: Terminal Equipment

I have set up a new estension as well as tried to register to an old extension trhat I was not registered to and I could not register. Specific message posted below from X-Pro. The very strange part is that my device that has been registered for days is still working.

I see thetre is a post on this topic in German as well, however I do not understand it.

RECEIVE TIME: 6162671
RECEIVE << 217.195.32.11:5060
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 10.240.132.34;branch=z9hG4bK6DC4E3156F224D6DA88257259AAEC90F;received=217.195.32.11;rport=50440
From: markosjal-204 <sip:markosjal-204@217.195.32.11>;tag=2216258729
To: markosjal-204 <sip:markosjal-204@217.195.32.11>;tag=as1e02e225
Call-ID: 0C4AB977F05D479DB69F7C197ACC255F@217.195.32.11
CSeq: 61266 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:markosjal-204@217.195.32.11>
Content-Length: 0

26.02.2006 22:11

Forum: Miscellaneous

so set up an FWD trunk and send the DID there!

24.02.2006 17:31

Forum: Providers

That was a ditto with Braintel.pk as well. All calls went to voice mail.

DB Error: syntax error 23.02.2006 08:22

Forum: Bugs

When I go to configure one trunk that I made the other day, I get the following error

DB Error: syntax error

It appears I can not delete it or do anything with it. Can an Admin delete it so I can at least try to recreate it?

trunk is

Trunk /2106796


Thanks

22.02.2006 05:56

Forum: Providers

Although the Italian service recommended as an alternative on this thread may use Port 5060, I have been anable to get a call from it! My phone rings when it is dialed and the caller keeps hearing the ring tone.

EDIT number 2
I prevously thought it did not work because it was g729 only , but turns out I found english documentation that I could understand better and found g771 is supported.

inbound routing or incoming calls? 19.02.2006 09:17

Forum: Miscellaneous

What is the difference between inbound routing and incoming calls? It seems that once I define something in incoming calls I can not undo it. Are they both necessary?

Distinctive Ring 19.02.2006 07:48

Forum: Feature Requests

Support for Distinctive ring might be a nice freeature to have especially for those who have DIDs from multiple countries. It is always nice to know what language to answer the phone in so you do not scare off prospective new clients.

I know that distinctive ring is implemented differently on some devices, however the most common seems to be the method that Sipura uses.

18.02.2006 02:55

Forum: Miscellaneous

Password field IS empty, although I have tried it with a Password as well, and even with a Password, it did not work

Also,when sending traffic to a SIP address it will likely be a server that I do not have an account on to register, however WILL accept SIP inbounds. I use FWD as an exampe, however FWD is not my interntion. It is probably I would want a few of these.


Can you give me an example of how I use the route to signify a SIP address?? Continuing with the 613 example would mean if I dial 613, it would route to sip:613@fwd.pulver.com (or sip/21488@fwd.pulver.com), however from the examples given, I do not get it.

Also I should clarify again that I DO NOT want a native bridge I want an External bridge (SIP redirect).


EDIT......


After posting above, I have now successfully made an outbound call. It seems deleting the route code several times and/or changing it, it finally worked.

Also, Is there a limitation of defining a single trunk? I have tried but it never seems to add it.

PLEASE help with sending traffic to a SIP address!

help me please! 17.02.2006 23:43

Forum: Miscellaneous

I want to achive a couple of things uing this service and would be willing to pay a reasonable fee to accomplish, however thus far I have had no luck.-


I have registered to the server and established a route to the PSTN , however the system askes me for a password even though I have left the password blank.

Secondly inbounds to not seem to be passing through either.

IOn addition and independent of the other itesms.

Can I register to an external server and do a simple SIP forward to a SIP address such as 613@fwd.pulver.com?

I believe in asterisk terms this is called an external bridge as opposed to a Native bridge. I do not see how I would differentiate between these two in your system.

17.02.2006 18:50

Forum: Providers

I believe you can also send those to sipphone , and they may not require the *

Also if you happen to be using Linksys or Sipura UAs try using the dialplan from www.sipbroker.com, however this will bypass the PBX functions.