RE: [Asterisk-Users] PRI for Data and Voice
One MORE Adtran Atlas 550. I have used it in service for over 3 years and it is ROCK SOLID!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Saturday, January 29, 2005 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI for Data and Voice On Sat, 29 Jan 2005, David Norton wrote: Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRI's of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Off the top of my head I can think of a few solutions: * Use a multiport T1 zapata card (TE405P or TE410P) and connect your systems this way: PSTN -PRI- Asterisk -PRI- Portmaster \ ---lan--- voip stuff With suitable parameters to the Dial application in Asterisk the forwarded calls will be passed transparently between the interfaces. This is similar to how we handle isdn data calls. * The zapata driver can handle digital but not analog ppp connections in the driver. If you wanted to you could use the above solution but have the Asterisk box handle the digital data calls. Not much is gained since you still need the Portmaster for the analog data calls. * Use a pri card with an on board DSP in the Asterisk box. The so called active isdn cards are usually so equipped. Cards in this category are the Eicon Diva Server T1/PRI cards (or the E1 equivalent) and probably more. I think they all interface to Asterisk via CAPI. * Buy a box with a dedicated box that does both VoIP and data call termination and interface to Asterisk via VoIP. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
Do you have a config sample on how to handle digital PPP sessions in Asterisk? On Sat, 29 Jan 2005 15:16:51 +0100 (CET), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 29 Jan 2005, David Norton wrote: Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRI's of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Off the top of my head I can think of a few solutions: * Use a multiport T1 zapata card (TE405P or TE410P) and connect your systems this way: PSTN -PRI- Asterisk -PRI- Portmaster \ ---lan--- voip stuff With suitable parameters to the Dial application in Asterisk the forwarded calls will be passed transparently between the interfaces. This is similar to how we handle isdn data calls. * The zapata driver can handle digital but not analog ppp connections in the driver. If you wanted to you could use the above solution but have the Asterisk box handle the digital data calls. Not much is gained since you still need the Portmaster for the analog data calls. * Use a pri card with an on board DSP in the Asterisk box. The so called active isdn cards are usually so equipped. Cards in this category are the Eicon Diva Server T1/PRI cards (or the E1 equivalent) and probably more. I think they all interface to Asterisk via CAPI. * Buy a box with a dedicated box that does both VoIP and data call termination and interface to Asterisk via VoIP. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
On Mon, 31 Jan 2005, Eric Bishop wrote: Do you have a config sample on how to handle digital PPP sessions in Asterisk? No, but there may be examples in the wiki: http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zapraspreview=3 http://www.digium.com/downloads/ppp.txt http://www.digium.com/downloads/hdlc.txt I think the last two are for permanent leased connections and possibly not what you are looking for. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
Dave, howzit You can use asterisk with a quad E1 card to divide your E1. So anyone who dials in using 1234 for example, route to your portmaster and anyone who dials in using 1235 use for IVR/voip, whatever. Good luck Regards Clive On 29 Jan 2005 at 15:11, David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated Regards David Norton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
On Sat, 29 Jan 2005, David Norton wrote: Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRI's of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Off the top of my head I can think of a few solutions: * Use a multiport T1 zapata card (TE405P or TE410P) and connect your systems this way: PSTN -PRI- Asterisk -PRI- Portmaster \ ---lan--- voip stuff With suitable parameters to the Dial application in Asterisk the forwarded calls will be passed transparently between the interfaces. This is similar to how we handle isdn data calls. * The zapata driver can handle digital but not analog ppp connections in the driver. If you wanted to you could use the above solution but have the Asterisk box handle the digital data calls. Not much is gained since you still need the Portmaster for the analog data calls. * Use a pri card with an on board DSP in the Asterisk box. The so called active isdn cards are usually so equipped. Cards in this category are the Eicon Diva Server T1/PRI cards (or the E1 equivalent) and probably more. I think they all interface to Asterisk via CAPI. * Buy a box with a dedicated box that does both VoIP and data call termination and interface to Asterisk via VoIP. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI for Data and Voice
David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated I betcha Sangoma has something that'd do this for you. They've been supporting T1 data on Linux for years, and they're recently added zapata to their list of open-source drivers. Give them a shout, they love this kind of stuff. Cheers, Jim -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI for Data and Voice
On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote: David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated I betcha Sangoma has something that'd do this for you. They've been supporting T1 data on Linux for years, and they're recently added zapata to their list of open-source drivers. Give them a shout, they love this kind of stuff. Of course when you go to using the Sangoma cards with asterisk, it appears you lose any extra functionality Sangoma built into the card. That isn't a bad thing, but it negates any benefit of longevity. As for the original posters question. The TE cards from Digium can take care of your ISDN dial ups by itself. Asterisk can't take care of your analog dialups yet. The first thing to know is that you are not splitting the PRI, you are routing calls. Until you get the setup messages, you don't know what is what. Then when you get it, the call could be on any of the B channels. But once you get it, you can determine by the phone number that was dialed how to route the call. You can assign a DID for your dialups and route it all to your portmaster through a separate span or assign different numbers for ISDN and analog dialups so only the modem users go to the portmaster while your ISDN users are handled on the asterisk machine. All others are voice and dealt with from inside asterisk. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
I met Sangoma guys this Tuesday, and got a AFT102 for evaluation. Right now I am in progress to develop a * channel driver for AFT10* devices. In that case you will have much more flexibility and to use all their API. Steven Critchfield wrote: On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote: David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated I betcha Sangoma has something that'd do this for you. They've been supporting T1 data on Linux for years, and they're recently added zapata to their list of open-source drivers. Give them a shout, they love this kind of stuff. Of course when you go to using the Sangoma cards with asterisk, it appears you lose any extra functionality Sangoma built into the card. That isn't a bad thing, but it negates any benefit of longevity. As for the original posters question. The TE cards from Digium can take care of your ISDN dial ups by itself. Asterisk can't take care of your analog dialups yet. The first thing to know is that you are not splitting the PRI, you are routing calls. Until you get the setup messages, you don't know what is what. Then when you get it, the call could be on any of the B channels. But once you get it, you can determine by the phone number that was dialed how to route the call. You can assign a DID for your dialups and route it all to your portmaster through a separate span or assign different numbers for ISDN and analog dialups so only the modem users go to the portmaster while your ISDN users are handled on the asterisk machine. All others are voice and dealt with from inside asterisk. -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
