Re: [asterisk-users] T.38 pass-through 488 handling problem
Klaus Darilion wrote: Steve Underwood schrieb: There seems to be a common misconception about 488. It represents an irrevocable failure of the call. Once a 488 is sent the call is essentially dead. A number of systems are able to continue beyond a 488, and allow further renegotation to another codec, but that it non-standard behaviour. The correct thing is to offer the options of T.38, u-law and A-law. If the other side can't do T.38, it should accept u-law or A-law. If it says 488, your dead. I tend to disagree. From RFC 3261, page 16: ...The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. I thought the same thing at one point, and kept complaining about things like Audiocodes, which just dump the call after sending a 488, and various other things which either dump the call or let the call continue but refuse to accept any further re-invites. Then someone pointed me to a later document that basically reverses what RFC3261 says. Right now I can find which document that was. Either way, its the real world that matters most. In the real world a lot of equipment will not behave well after a 488. We've had lots of experience with this in T.38 testing. I'm chasing this very issue with Audiocodes on a Mediant 1000 right now. Is there any way to work around this? Is there a way in asterisk to intercept the 488/486 and suppress it's behavior to dump the call? peter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI
Using a T1/E1 ISDN interface it's somewhat trivial. In zapata's conf: group=0,11 context=from-pbx-custom switchtype = national signalling = pri_net pridialplan=national prilocaldialplan=national channel = 1-23 group= context=default Note the pri_net for signalling. I have several PRI spans running this way to PBXs. Then configure your dialplan to match the remote site's extensions to use the right trunk interface. On 5/16/08 9:48 AM, Al Baker [EMAIL PROTECTED] wrote: This is 'basically' a tie-line between the boxes. Yes - it is done all the time between PBX's. You are basically nailing up a circut between the boxes. It could be a simple as a simple POTS leased line or a multi-t1 bundle between them. How it is physically done with DIGIUM's boards under * ? Someone else will have to answer that Pascal Maugeri wrote: Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when they have to interconnect. Now, might it be possible to connect directly (A) and (S) using their PSTN interfaces without having to go through to my Telco ?! Does it make sense ? Is it technically feasible ? I guess that the Telco network is providing routing, number assignation, etc. and it sounds pointless to do this. Nevertheless could you confirm it is possible/impossible and why ? Is there a better way to do that ? Thanks in advance, Pascal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending caller name out PRI?
On 5/1/08 4:17 PM, Peter A Eisch [EMAIL PROTECTED] wrote:
Is there a way to delay (or resend) the name much like the carrier does?
This would then be closer to what the carrier does (as in how I need to
have a Wait(1) before using ${CALLERID(name)}). This assumes that it's a
timing issue I guess.
[following up to myself]
I've debugged on the PRI from a telco carrier and they never send the name
message in the setup with the ANI and DID -- it always comes after.
Looking more at the message as zaptel writes it to the wire, if the logging
is right, the first datum in the setup message is the name. Is there a way
I can get the name to come after the DID and ANI even in the same message?
My current guess is that the PBX discards the name because there's no call
already existing.
Is there a syntax that I can use to copy the carrier behavior?
Thanks!
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Re: [asterisk-users] AsteriskNOW and IE
It might be worth pointing out too that you can't use Firefox 2.x on OS X. I had to solely use Ff on Win2k to do anything at all with the GUI. I hadn't tried linux. YMMV. peter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting callerid across servers
On 3/11/08 2:53 PM, Niles Ingalls [EMAIL PROTECTED] wrote: On Mar 11, 2008, at 3:25 PM, Jerry Geis wrote: I have a situation when a T1/PRI line comes into box 1 then uses SIP over to box 2 and all my phones are on box 2. if the person is not at their desk on ring no answer I am calling their cell phone which places the call back over SIP to box 1 and out the T1 . How can I setup this configuration so the original caller ID will show up on the cell phone. Thanks, Jerry Jerry, What CID are you expecting to show on the cell phone? Based on what information you have provided, the original call is coming outside of your system, and you will not be able to duplicate their CID when you pass the call to your users cell phone. You can always screen the call though, allowing the recipient to know who is calling them. Niles This can vary on a PRI from provider to provider. I'm currently switching vendors because the current one will only take specific CID info. The new vendor will allow any CID info (on or off their network) to be presented. The old vendor keeps promising me that they have an update for their system which will allow them to take any -- but their time ran out. peter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
