RE: [Asterisk-Users] Dell Hardware
I'm using a Dell GX270 with a single TE110P, no problems here. Of course I had to take off the pci aluminum card holder thingy to fit in the half height case, but it works great. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thibault Lamy Sent: Friday, July 22, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Hardware We are using a Dell PE SC1420 as asterisk server with one Beronet QuadBRI card (bristuff) and one Digium TE110p and it works well. No IRQ conflict. Thib. Elio Rojano wrote: DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ Bruno De Luca escribió: We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ 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] long pause on dialing..
I'm just using some generic GE phone (analog) connected to a linksys pap2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, July 18, 2005 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] long pause on dialing.. Sounds like its on the phone. What type of phone are you using. - Original Message - From: Giorgio Incantalupo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 18, 2005 9:40 AM Subject: Re: [Asterisk-Users] long pause on dialing.. Hi, it is hard to answer without the right piece of extensions.conf but remember there is a digit timeout in Asterisk: you enter 9 digits but in the dialplan there is a match for 10 digits so how can Asterisk know if you want to call the 9-digits number or the 10-digits? After 9 digits it waits for a while...if another digit is dialed then it can call the 10-digit number otherwise it calls the 9-digits number. You can lower Asterisk digit timeout but remember that not all users are so fast to dial... Giorgio. Goolsby, Daniel S (Daniel) wrote: I have an Asterisk setup with AMP installed. I have phone extensions from 7000 to 7010. I experience long delays when dialing a 9 digit number as opposed to a 10-digit number. How do you get around not having to press the # key to speed up the dialing process? For any length phone number for that matter-- like dialing another extension. If I dial 7005, I'll have to wait a while.. but it's instant when I press the # key. Daniel ___ 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 ___ 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
[Asterisk-Users] massive outbound calling...
Does anyone know what kind of limitations asterisk has when it comes to massive outbound dialing.. i.e. how many sip/iax phones could be dialed at the same time-- and if someone answered, play a .wav file? Or outbound throughput on zaptel devices? Say if I had a dual xeon with 2 quad t1 cards, hooked up to a 100mbit lan. Anyone know how many it could actually sustain w/o the voice file being distorted on playback? Daniel ___ 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] long pause on dialing..
I have an Asterisk setup with AMP installed. I have phone extensions from 7000 to 7010. I experience long delays when dialing a 9 digit number as opposed to a 10-digit number. How do you get around not having to press the # key to speed up the dialing process? For any length phone number for that matter-- like dialing another extension. If I dial 7005, I'll have to wait a while.. but it's instant when I press the # key. Daniel ___ 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] Extension Configuration Best Practice
Could you just configure the extention to be a ring group instead of an actual extention, or ring queue.. then have his phone/laptop log in whenever he's at the office/coffee shop? I know AMP has the functionality, but I haven't gone behind the scenes and looked at the sip.conf or extensions.conf to see what the script or macro is doing in a ring group/queue. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, June 21, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Extension Configuration Best Practice Ok, so how are you guys coping with scenarios like this? Managers working in the office during the day or mid day and then in the afternoon, working remotely using their laptops? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:20 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. | |- Joshua Colp. | | |On 6/21/05 9:39 AM, Rich Adamson [EMAIL PROTECTED] wrote: | | I would like to hear tips and tricks on extention config best | practices, for example, naming, etc. and most of all, how to deal | with extention that have a full time hardphone configured and | assigned and then a softphone connecting to the same extention, for | example, one employee has its hardphone on the office but sometimes | when he travel, he uses his softphone to work with, what |happens when | two phones have the same user id and connect to the same asterisk? | How are calls routed or how to handle this kind of scenarios. | | In general terms and without being able to see how the extension is | defined in sip.conf, the last phone to register with * will get the | call. | | Assuming both the hard and soft phones register every hour, it is | entirely possible the hard phone will get the call for the first 30 | minutes and the soft phone for the next 30 minutes. | | | ___ | 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 | ___ 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
[Asterisk-Users] Wildcard TE110p initial setup
Im not overly new to asterisk, but I am very new to the Digium card / Telco aspect of it. I was able to get a TE110p card, installed.. and hooked a t1 up to it. Im using zttool to just look at some output. Here is what it looks like when I just load it up: Current Alarms: No alarms. â â â â Sync Source: Internally clocked â â â â â IRQ Misses: 0 â â â â â Bipolar Viol: 0 â â â â â Tx/Rx Levels: 0/ 0 â â â â â Total/Conf/Act: 24/ 24/ 0 â â â â â 112 â â â â â 123456789012345678901234 â Back â â â â â â TxA â â â â â TxB â â â â â TxC â # â â â TxD â â â â â â â â RxA â Loop â â â â â RxB â â â â RxC â â â â RxD â If I make a call to one of the phone numbers in my block, the zttool output goes to this: â Current Alarms: No alarms. â â â â Sync Source: Internally clocked â â â â â IRQ Misses: 0 â â â â â Bipolar Viol: 0 â â â â â Tx/Rx Levels: 0/ 0 â â â â â Total/Conf/Act: 24/ 24/ 1 â â â â â 112 â â â â â 123456789012345678901234 â Back â â â â â â TxA â â â â â TxB â â â â â TxC â # â â â TxD â â â â â â â â RxA 1000 â Loop â â â â â RxB 1000 â â â â RxC â â â â RxD â Channel 1s RxA and RxB goes to 1, while nothing else happens. Im running somewhat of a latest build, with AMP installed. I see that its configured that any incoming call is in the from-pstn, but I have yet to get a busy signal, or any response from the card. What I would like to know is it my card, software, or t1? Ill give up just about any information asked for, if someone could help me. Daniel ___ 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
