Re: [asterisk-users] Transfer feature - howto?
Eric ManxPower Wieling wrote: I don't know if this is even possible. I might be totally wrong but once this call is on the cell network, how are you gonna communicate with asterisk?? From what I understand, while the voice (RTP) traffic still travels through asterisk, You have no access to any kind of signalling. Please correct me if I'm way off base here, anyone. You are offbase. Even with reinvites the SIP SIGNALING will continue going thru Asterisk. Ok. Thanks! So how does one go about getting asterisk to recognize DTMF in this situation? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE IP Phone
bilal ghayyad wrote: Hi List; I am looking to use an good IP Phone working with Asterisk and work based on PoE (so it takes the power via the ethernet cable, no need to connect for it separated power adaptor). Can someone advise me for good one? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We use cisco 7960's which are PoE. They work pretty good (Decent speakerphones). They're a little on the high side and I wish the softkeys were more configurable ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange problem(Munin-node-1.2.4-7)
Crazy Boy wrote: Hi, Sorry to post this in this forum. I am new to Trixbox. When I am trying to install Trixbox, I am facing this problem. First I have installed Trixbox ISO image file from a CD. When its rebooting and Asterisk is installing, it is got stucked near this below point: Munin-1.2.4-7 Preparing package for installation... 0:group munin already present 0:user munin already present Munin-node-1.2.4-7 and stopped at this moment. Why this is happening? I tried to installed Trixbox 3 times. But, I faced this problem evertime. Please tell me the problem. Looking forward to your response. Thank you. Regards, Chandra. http://lists.digium.com/mailman/listinfo/asterisk-users http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com Chandra, You might have more luck asking this in the trixbox forum. I received the same problem. I think all I did was power off the box and reboot and it went all the way to the end of the install. I don't know why this happens, sorry I'm not of more help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need USA DID + trunk provider
R.R. Libera wrote: Hello, I need an USA DID + 15 b-channels. The only option I already have is OpenVox and I want to see some alternatives. Sound quality is my priority. Thanks in advance. R.R Libera ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We've been using voipstreet. Excellent sound quality and unlimited channels on incoming DID's. A little bit on the expensive side as DID's are $3.00 a month + 1.4 cents a minute. The worst part is that they charge minimum of 1 minute a call. Their level of support, and availability is the best bar non however ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer feature - howto?
Mike wrote: Hi, My setup is the following: Voip provider---(SIP DID)---Asterisk box(SIP through a termination provider)---multiple cell phones. The cell phones each have their extension (201,202,203,204) and I'd like to be able to have them transfer a call to somebody else. Ex: Prospect calls extension 201, talks to the salesgy, who forwards him to the tech guru somehow. My guess is I have to use the transfer feature found in feature.conf. I tried, no success. What Wiki page do I need to look at to get details on this? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I don't know if this is even possible. I might be totally wrong but once this call is on the cell network, how are you gonna communicate with asterisk?? From what I understand, while the voice (RTP) traffic still travels through asterisk, You have no access to any kind of signalling. Please correct me if I'm way off base here, anyone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer feature - howto?
Do you know for a fact that inband DTMF is being procesed by Asterisk when the call hits the cell? Well it seems I'm wrong but how do you setup asterisk to process inband dtmf? I dial the cell phone with the 't' in the dial string. When I hit anything from my cell phone, asterisk doesn't seem to process this. Is there a setting to enable inband dtmf? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phones
bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any phone supporting SIP or IAX are good choices for asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asteisk-users]USA DID + trunk
R.R. Libera wrote: Hello Steve, Thanks for recommend VoipStreet to me; their product look good. Could you please tell me more about it? Could you please give, approximately, the REAL amount of money for a certain amount of minutes? (based on your experience). Somebody offer this flat rate: $15.00 for the DID. $17.00 for each channel monthly $10.00 for each channel, one-time install fee This is for typical business or home use. Is that more expensive or cheaper than VoipStreet? Thanks in advance. R.R. Libera It's real hard to say without having at least some idea of your monthly usage. What we pay for voipstreet is $3.00 a month for the DID + 1.4 cents a minutes. The only caveat is if a call lasts 30 seconds. They still charge you for the full first minute (1.4 cents). After 1 minute it goes to 6 second billing. That gets expensive when you're doing a lot of inbound minutes. We were looking at voxbone and even though they have unlimited inbound minutes for $5.00 a month they limit you to two channels. Extra channels are 22.00 a piece and you have to get 10 minimum. That's where they nail you. We might still use them for some of our numbers that don't have a lot of simultaneous calls. $15.00 for the did seems high. the $17.00 for the channel is pretty good though. Do you absolutely need multiple channels per DID? If you can get by with 2 channels things would be a lot easier. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asteisk-users]USA DID + trunk
R.R. Libera wrote: Thanks Steve for your timely response. I really needs 15 channels an one DID cause I can´t make my customers to remember 7 PSTN numbers in order they can reach me. That´s why I must have a DID + trunk. Just for getting an idea, we need to give 2300 minutes of support in order to make some difference. So, that´s the MINIMUN I can make in order to get a bussines. Putting numbers clearly we have: *VoxBone* Install: 15 channels x 36 USD = 540 usd Monthly Charge: 15 channels x 22 USD = 330 usd I REALLY don't know why they have to charge so much for additional channels. I guess that's what makes sense for them. I would understand if an additional channel cost the same as a DID but not 4 times as much. I don't get it. *MetroTel *Install: 15 channels x 10 USD = 150 USD Monthly Charge: 15 channels x 17 USD + 15 USD = 270 USD *VoipStreet *Install: 3 USD Monthly Charge: 2.95 usd + (2300) x 0.0141 = 35.38 USD * Of course, that last formula it is not so close to the real life, but things seems to be Ok with VoipStreet. The only caveat is the surprise factor. This of course sounds good. But you would be surprised how quick you rack up 2300 minutes. If you're running a support line there is NO WAY you're going to be at 2300 minutes. You might also need to triple check with voipstreet to ENSURE they can handle 15 simultaneous calls on one DID. I don´t really understand you when you said: They still charge you for the full first minute (1.4 cents). After 1 minute it goes to 6 second billing. I mean that AFTER the first minute of the call they charge you in 6 second increments. This means that if someone calls you for 1 minute and 12 seconds they charge you: 1.41 cents for the first minute + 0.0011 * 2 for the next 12 seconds. If someone calls you for 30 seconds though, you still pay 1.4 cents. This doesn't matter if most of your calls are 1 minute. but if you have a lot of short calls it really sucks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements
I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the default 0.01 for the Blocking. It worked out to 264 kbps. Does this sound reasonable? If so great! A business DSL could support this. Comments welcome! Sounds about right to me. 8kbps + overhead. Transcoding 20 simultaneous channels requires a relatively hefty machine I think ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bandwidth usage
Hello, I thought I'd ask this, just in case I'm wrong. We're trying to set up 'remote' users via asterisk. Basically all there is to this is asterisk forwarding a DID to a cell phone. My question is this: Is there any possible way for our local asterisk box to setup the connection and the drop out so that our bandwidth isn't being used for the call? I really don't think this is at all possible but I thought I would double check. Perhaps with the cooperation of your DID provider? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to asterisk DID extentions
Matt wrote: We have two * boxen linked with an IAX trunk they are on different networks. Calling from any ext to any other remote ext on either box works fine.Outgoing calls from either box also works fine. Problem when dialing in from DID (SIP or IAX provider) number, the caller cant transfer to any ext that is not on the local box. How can I make * aware of the other ext on the remote box so the DID caller can access them like he can with the local box? local ext are 8xx remote ext are 9xx thanks -Matt What about DUNDi? I'm about to roll this out in the next few days, I'll let you know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 voip to failover pri
stan ford wrote: I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to have a PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? is anyone out there, using a VOIP only with no failover? We're using VOIP only, no failover. Furthermore we're using it on a cable internet connection. We have a cheap dsl connection for backups. It's been up for about 2 months now and has only been out twice for a small period of time. When that happens the DSL takes over. I don't pretend that this is in anyways comparable to PSTN service but it works pretty well for us. We have three locations. Two of which are set up the same way, the third just has 3 stations and just registers with one of the asterisk boxes at the other locations. I think when you're talking enterprise you definitely want to go with a t1 or two t1's for backup. (I don't really understand how a PRI gives you more reliability than a T circuit. They run over the same copper don't they??) For our purposes however (and I'd like to think I speak for a lot of mid size businesses with 50 employees) our setup works wonderful. It costs us about $600 all in all (internet access + VOIP) and that's a FAR cry from what we were paying through Covad before. Of course there always will be exceptions (People that need 100% guaranteed uptime), but for the size of our business this works. The only part that REALLY concerns me is our DID's. If our DID provider ever goes down we are screwed. Anyone know of any failsafes for THIS? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn failback
B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner. I have yet to come up with any way that failure of our DID provider could be handled. We've had to do a fair bit of shopping around before we found a voip provider that has been reliable (Voipstreet). They've had very good service to this point (about 3 months now. I don't know what we would do if they went out. A PRI is just far to expensive to consider for only backup purposes. DID's have been somewhat of a problem for us. There aren't that many providers and those that exist are more expensive than I thought it would be (or if they're cheap their service is so unreliable as to be worthless). Maybe someone here can give me an idea of how they do their DID's ? We've tried didx.org as well but most of the numbers we've gotten from them have horrible audio problems. Maybe someone knows of a super reliable provider with unlimited incoming/ 2 channels for about 10$ a month? Termination is no problem. Lots of providers and the dialplan will automatically failover if one provider doesn't work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks everyone for all the info. I'm going to assume this is normal for this phone and doesn't adversely affect performance. I've installed these on several different asterisk systems (including 1.4) and it's all the same. Anyone running sip firmware 8.4 know if this is 'fixed' ? Again thanks for all the help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
All my Cisco phones show less than 75ms except for one (mine of course). I do have a switch in my cube that I use for extra ports and that's the only real difference. Do you have anything plugged into the extra network port on the phone? Yes, I have workstations plugged into the extra ports on some of the phones - Doesn't seem to make a difference What's in between your phone and the asterisk server? My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
Mailing List wrote: My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. basically nothing is wrong. I have a 3com switch in front of the one phone that reports the large time. Now I'm thinking it has something specifically to do with 3com switches and these phones. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That's a possibility but I took one of these phones home and pointed it at my own asterisk system and it reports ~ the same. What make and model 3com switch are you using? what does the phone in your office report? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Very high ping times from 7960 phones
I've asked this here before and never really got a response, so I'll try again :) I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. Does anyone have any idea what might be causing this? I thought that it might just be a 'reporting' issue but there is definite latency there when I do an echo test. I'm running cisco sip firmware 8.2 on all the phones. Thanks for any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom high SIP ping time
Mike Hammett wrote: I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify was returning ping times anywhere from 20 to 70 ms over a sparsely used LAN. Command prompt (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. Any ideas? Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm having much the same issues only I'm using Cisco 7960 phones. When I do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone on the same network (x-lite), is reporting times of 4 ms. Related to this (I think), I'm getting audio issues. The person being called can hear the caller fine but the callee's voice drops in and out excessively. I have qualify set to yes in the sip definitions for all the clients (Including the soft phone). Does anyone know what is causing this. I'm not aware what the sip ping times were earlier, but the audio issues seemed to have started spontaneously. Anyone have any idea regarding this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom high SIP ping time
Steven Ringwald wrote: Steve Glaus wrote: Mike Hammett wrote: I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify was returning ping times anywhere from 20 to 70 ms over a sparsely used LAN. Command prompt (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. I'm having much the same issues only I'm using Cisco 7960 phones. When I do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone on the same network (x-lite), is reporting times of 4 ms. Related to this (I think), I'm getting audio issues. The person being called can hear the caller fine but the callee's voice drops in and out excessively. I have qualify set to yes in the sip definitions for all the clients (Including the soft phone). Does anyone know what is causing this. I'm not aware what the sip ping times were earlier, but the audio issues seemed to have started spontaneously. Anyone have any idea regarding this? What codecs are you using? I have noticed that g729, for some reason, adds a lot of latency to the phone. Running on uLaw, however, I get times from sip show peers of around 5-14ms. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm using ulaw all the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom high SIP ping time
Mike Fedyk wrote: Steve Glaus wrote: Mike Hammett wrote: (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. Any ideas? I'm having much the same issues only I'm using Cisco 7960 phones. When I do a 'sip show peers' I'm getting times in excess of 300ms. A soft phone on the same network (x-lite), is reporting times of 4 ms. Related to this (I think), I'm getting audio issues. The person being called can hear the caller fine but the callee's voice drops in and out excessively. I have qualify set to yes in the sip definitions for all the clients (Including the soft phone). Does anyone know what is causing this. I'm not aware what the sip ping times were earlier, but the audio issues seemed to have started spontaneously. Do you have any problems when there are a low number of concurrent calls? Do you ever get any messages saying the phones are unreachable? Or lagged? What kind of Internet connection do you have? Do you have any problems with calls between phones on the same network (no routers in between)? What model of switches do you have? What model of Internet router do you have? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you have any problems when there are a low number of concurrent calls? Yes, from what I've seen and tested, it doesn't make a difference how many calls there are. Do you ever get any messages saying the phones are unreachable? Or lagged? Yes, a phone not being used often enough will often report 'UNREACHABLE' When a call is made that changes What kind of Internet connection do you have? We have a synchronous T1 connection. The phones run on the same network as data. Do you have any problems with calls between phones on the same network (no routers in between)? There are some static issues but nothing like outside calling were the person being called drops off for 10 seconds at a time. Switches used are 3com Baselines 2024 What model of Internet router do you have? Cisco 1700 router My main concern is the fact that the soft-phone is reporting only 4ms of delay and the cisco phones are reporting 200. All the time, regardless of network bandwidth. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 BLA
While I'm frantically scouring this list, does anyone have any information about getting BLA (busy line appearance) working on Cisco 7960? The last I heard was that this was unsupported in Cisco's SIP firmware ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
