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sky
Grünschnabel


Registration Date: 03.07.2006
Posts: 65

SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

When My softphones (sjphone, xlite, ...) or my E60/61/70 are registered at pbxes... then i can't call anyone anymore that is not in my extensions list or in my outbound routing list.
Eg a "someone@telefoon.in" will be not possible.

I tried making local mirror extensions eg ;
extension 200 -> sip:someone@telefoon.in
but that didn't seem to work?

Currently I've created a default outbound profile with voipbuster on my E70 which allows sip uri...

on http://blyon.com/sip_uri/ someone explains how to get sip uri to work on an asterisk?

08.11.2006 15:16 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
Dia
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Well I have what you describe working since the early days of PBXes, but for outgoing calls only. That is, I can call extention xxxx from another real extesion (one that registers to PBXes) and get to a SIP URI.

To get this to work, I created a SIP extension, and in the "Device Options" put in the "Dial" field: SIP/username@sip.someproxy.com

Don't leave the password field blank as on all the other SIP extensions.

You need to ensure though, that the proxy accepts SIP inbound traffic, since some providers block it. e.g. Vonage, Stanaphone etc.

Let me know if this works for you, if not, I will try to help you troubleshoot it.

08.11.2006 23:03 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
sky
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I've read about that and configured the following
extension 53 with sip/someone@voipbuster.com
extension 530 with sip/someone@pbxes.org
extension 693 with sip/someone@telefoon.in
extension 739 which is registered with my sipura3000

if I call 739; my caller&called extension goes red.
if I call the others the called extension remains green (not grey!)
they turn up in the logs...
but I don't have any sound while making the call;
the called person has no extensions ringing nor has he got logs of my call attempt in his pbxes account...

so company X has a pbxes.org account
and company Y has a pbxes.org account

people of company X will get a 2xx at company Y
but somehow they aren't reacheable?

09.11.2006 11:13 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
sky
Grünschnabel


Registration Date: 03.07.2006
Posts: 65

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I call this more the "speeddial" feature like you can find on voxalot... eg make an extension that you can "dial" from an ip phone or phone behind an ata or via a disa or sipbroker dialin number...

but if someone calls me on my sip-enabled phone; they leave a callerid... "somebody" something@somewhere

now I could look up the "somebody" in my contactlist and hopefully he's got an extension in my pbxes... or an enum-field;

but as long as my softphone is registered at pbxes...it will not allow me to dial "something@somehwere" or it will send that to pbxes...

also the "somewhere" is replaced by the ip of pbxes while the "something" stays... which makes me have to look at the pbxes logs to know which incoming trunk is used...and append that sip ip to the "somewhere" to be able to create a callback extension...

09.11.2006 11:17 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
Dia
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From what you described all of the extensions should work (provided they accept inbound SIP calls), except this one: extension 530 with sip/someone@pbxes.org

This should be implemented through the PBXes to PBXes calling feature. Also if you want to reach a paricular extension you should create an Inbound Route to direct the call to the desired extension.

Pascal can you please comment on the second message, since if I understand it well, it means outgoing calls based on a SIP URI.

10.11.2006 05:52 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
sky
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could you tell me where to implement this feature " PBXes to PBXes calling feature." ?

10.11.2006 12:11 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
sky
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oki my contact just had his phone ringing so that one seems to work...
I've had to disable "audio bypass" for his extension and then we could discuss...


Now hopefully "sip uri" calling would be introduced as well...
I'd like not to start entering everybody in my pbx and then adding their shortcodes into my contacts list... (it's feaseable but ...) -> Feature Request

And hopefully when they call me I get their "sip uri" correct and not replaced with the ip of pbxes? -> Bug

10.11.2006 15:08 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
i-p
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Registration Date: 14.01.2006
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Lampe RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

PBX-to-PBX calling is described in the News section of the forum.

We have allowed URI calling as per your request.

Concerning Caller ID:

  • For PBX-to-PBX, URI and ENUM calling (but not for what you called speeddial and also not for calling thru classical extensions) we changed the Caller ID to be account-extension@pbxes.org.
  • For incoming calls the trunk's IP cannot be shown. The Caller ID is not designed to include the IP address of the caller.

This post has been edited 1 time(s), it was last edited by i-p on 29.11.2006 at 04:25.

10.11.2006 19:14 i-p-tel is offline Search for Posts by i-p-tel Add i-p-tel to your Buddy List
sky
Grünschnabel


Registration Date: 03.07.2006
Posts: 65

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

URI testing:

When someone calls skynetbbs-739@pbxes.org from within his pbx... all his extensions are ringing (eg His incoming calls config is found/used which makes all his extensions ring)
My 739 extension is not ringing !

When he creates a local extension 739 which points to skynetbbs-739@pbxes.org then it works...


When he calls bennydm-300@pbxes.org he expects extension 300 to be calling... but it's ringgroup 1 that is ringing (which doesn't hold extension 300)... -> incoming call routing...

isn't it normal that
bennydm@pbxes.org should follow incoming routing;
but when someone deliberately mentions the extension it should try that extension (and if *92 is used follow that route...


Albeit thnx for making it work!

11.11.2006 13:10 skynetbbs is offline Search for Posts by skynetbbs Add skynetbbs to your Buddy List
Dia
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Outgoing SIP URI calls should be formed into a valid URI string, such as name@FQDN or email@xxx.xxx.xxx.xxx, while inbound URI strings should be in the form AccountName-Ext#@pbxes.org or AccountName-RG#@pbxes.org.

In the case of an inbound SIP URI call, if the URI conforms to a valid pattern, the call should be able to reach the extension or ring group when a specific Inbound Route has been set up. In the case where the inbound SIP URI contains a valid AccountName but is not followed by a valid Ext# or RG# then it should be send either to the default extension or ring group set in Incoming Calls.

So after the theory lesson, let's get back to the question at hand.

skynetbbs have you tried an inbound SIP URI call from another system (not another PBXes account) to reach a particular extension?

From what you describe bennydm's default destination in Incoming Calls is RingGroup 1, since there in no Inbound Route setup to direct the call to ext 300.

11.11.2006 20:10 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
dor
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Registration Date: 20.03.2008
Posts: 125

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I tried to do something similar, but unsuccessfully.
I created inbound route user-abcd@pbxes.org forwarded to VM, and tried to dial this SIP URI from X-Lite registered at PBXes SIP extension. Got a error message saying that number cannot be dialed.
When I registered that X-Lite client with non-PBXes account, I could type user-abcd@pbxes.org and get connected with no problem.
So, what went wrong when my client was registered at PBXes extension?

12.04.2008 03:18 doronin is offline Search for Posts by doronin Add doronin to your Buddy List
sof
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Registration Date: 13.05.2009
Posts: 14

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Hi, I am tying to call SIP URI number@sip.mydivert.com but pbxes hang up call.

I tried following way to call SIP URI

sip/number@sip.mydivert.com
SIP/number@sip.mydivert.com
number@sip.mydivert.com

Can you please let me know how can I divert call to SIP URI. Thanks

04.06.2010 17:23 softphone is offline Search for Posts by softphone Add softphone to your Buddy List
baz
Grünschnabel


Registration Date: 20.12.2009
Posts: 138

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Just to bump this really old thread but I'm having trouble too. I have a trunk, that goes to a ring group, that has one extension in which calls the sip uri. it hangs up immediately. I can call the SIP uri from my softphone fine.

Aug 25 11:05:33 VERBOSE[76241] logger.c: -- Called numberxxx@sip.gradwell.net
Aug 25 11:05:34 VERBOSE[66817] chan_sip.c: SIP response 100 to standard invite
Aug 25 11:05:34 VERBOSE[66817] chan_sip.c: SIP response 407 to standard invite
Aug 25 11:05:34 NOTICE[66817] chan_sip.c: Failed to authenticate on INVITE to '"My CID" <sip:mycidxxx@88.198.69.250:27504>;tag=as1d6c5d5a'
Aug 25 11:05:34 VERBOSE[76241] logger.c: -- SIP/sip.gradwell.net-71c5 is circuit-busy


edit:
to be clear numberxx is just a number. I think that pbxes is trying to dial the number using sip.gradwell.net as a proxy, rather than assuming it's on that system. If that makes sense...?

This post has been edited 1 time(s), it was last edited by baz on 25.08.2012 at 12:29.

25.08.2012 12:07 bazmercer is offline Search for Posts by bazmercer Add bazmercer to your Buddy List
i-p
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RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Calling numbers at SIP URIs e.g. ###@dom.tld does not work, because destination numbers are picked up and then routed by outbound dial rules.

You can try to exclude the number from your dial patterns defined in outbound routing.

Calling alphanumeric abc@dom.tld should be no problem.

27.08.2012 14:22 i-p-tel is offline Search for Posts by i-p-tel Add i-p-tel to your Buddy List
baz
Grünschnabel


Registration Date: 20.12.2009
Posts: 138

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Hi,

Yeah I thought that was the problem. I can dial it directly from my softphone though. I can't exclude it from any dial patterns with any ease as it's a regular landline number here in the uk. Any other ideas?

29.08.2012 09:02 bazmercer is offline Search for Posts by bazmercer Add bazmercer to your Buddy List
idd
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Registration Date: 21.11.2011
Posts: 3

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Zitat:
Originally posted by Diafora
Outgoing SIP URI calls should be formed into a valid URI string, such as name@FQDN or email@xxx.xxx.xxx.xxx, while inbound URI strings should be in the form AccountName-Ext#@pbxes.org or AccountName-RG#@pbxes.org.

In the case of an inbound SIP URI call, if the URI conforms to a valid pattern, the call should be able to reach the extension or ring group when a specific Inbound Route has been set up. In the case where the inbound SIP URI contains a valid AccountName but is not followed by a valid Ext# or RG# then it should be send either to the default extension or ring group set in Incoming Calls.

So after the theory lesson, let's get back to the question at hand.

skynetbbs have you tried an inbound SIP URI call from another system (not another PBXes account) to reach a particular extension?

From what you describe bennydm's default destination in Incoming Calls is RingGroup 1, since there in no Inbound Route setup to direct the call to ext 300.

30.08.2012 12:25 idd1717 is offline Search for Posts by idd1717 Add idd1717 to your Buddy List
neu
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Registration Date: 12.07.2012
Posts: 8

RE: SIP URI calling Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

"Calling numbers at SIP URIs e.g. ###@dom.tld does not work, because destination numbers are picked up and then routed by outbound dial rules.

You can try to exclude the number from your dial patterns defined in outbound routing.

Calling alphanumeric abc@dom.tld should be no problem."

But why? I have never come across a SIP URI that is not number@xxx.com

It seems a HUGE bug that the code assumes that a user that dials 1234@sip.org really meant to dial 1234.

01.10.2012 23:17 neuroscot is offline Search for Posts by neuroscot Add neuroscot to your Buddy List
 
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