Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
Yeah but it appears that Teliax just charges you if cross that limit for every minute you go over. So it's not a soft limit, that is just marketing spin. It's a hard limit. Be so much nicer if Starbucks were to use small, medium and large and ITSPs would just advertise fixed minute plans as cap'd minute plans. Never ever use the term unlimited unless you mean it. - Dustin - Rusty Dekema wrote: At least the soft limit is explicitly published (X Minutes) as opposed to most companies' policy of There is a soft limit, and we will not tell you what it is, but if you reach or exceed it we will [charge you $100/day | terminate your service | switch you to a more expensive plan without notice]. I am sort-of searching for a new SIP/IAX trunk provider and I would much rather have a policy like Teliax's than the others. -Rusty On 11/12/05, *Dane Reugger* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Teliax looks good - not comfortable with the soft limits but love the free setup! [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Have you looked into teliax? 4 simultaneous calls on a bus plan is pretty good for less than $50/mo. And I cannot complain about the quality or the support. Greg -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Saul Diaz Sent: Friday, November 11, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline Julio Arruda wrote: I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Broadvoice only allows only the normal 3 way calling so is 2 channels for # about BV i got a lot of water under the bridge every works ok supper ok for times. then BV brokes without you make a single change in your asterisk server and stop working.. if u call support you are the guy with the problem.. yes BV
Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
I was wondering about inbound. I got my first bill and got billed for extra minutes even though a CDR download showed inbound calls totals were under the softcap. I open a case and got NO response. I was hoping this was billing mistake but now I am starting believe Teliax is misrepresenting there services on a whole different level. If they are really calculating the softcap on inbound and outbound their webpage could only be treated as false advertising. - Dustin - Paul wrote: The softcap seems to include inbound minutes. Looks to me like it is better to get your incoming DID's from one provider and deal with termination separately. If someone like me offered unlimited consulting ** for $150/month but then said there was a ** softcap of 2 hours I would be called all kinds of names. But if I state my true opinion of what teliax is doing here people will jump all over my case. Look at it and smell it. Don't come whining to the lists about it after you sign up and then figure out that 1500 minutes/month is an average 50 minutes/day. Compare them to vonage. I and a few people I know have averaged about 1500 a month both inbound and outbound on vonage residential accounts for over a year now. We still have the same accounts and no extra charges, warnings or threats. In fact vonage replaced the original motorola ata with a linksys router free. I just told them how I was activating my friend using the same network I had the old ata on and noticed a great difference in voice quality. They gave me an RMA number and shipped the new unit right out. Dustin Goodwin wrote: Yeah but it appears that Teliax just charges you if cross that limit for every minute you go over. So it's not a soft limit, that is just marketing spin. It's a hard limit. Be so much nicer if Starbucks were to use small, medium and large and ITSPs would just advertise fixed minute plans as cap'd minute plans. Never ever use the term unlimited unless you mean it. - Dustin - Rusty Dekema wrote: At least the soft limit is explicitly published (X Minutes) as opposed to most companies' policy of There is a soft limit, and we will not tell you what it is, but if you reach or exceed it we will [charge you $100/day | terminate your service | switch you to a more expensive plan without notice]. I am sort-of searching for a new SIP/IAX trunk provider and I would much rather have a policy like Teliax's than the others. -Rusty On 11/12/05, *Dane Reugger* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Teliax looks good - not comfortable with the soft limits but love the free setup! [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Have you looked into teliax? 4 simultaneous calls on a bus plan is pretty good for less than $50/mo. And I cannot complain about the quality or the support. Greg -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Saul Diaz Sent: Friday, November 11, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline Julio Arruda wrote: I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number
[Asterisk-Users] sched.c: Attempted to delete nonexistent schedule entry
Is anyone else having all IAX peers die right after receiving this in the log? I have CVSHEAD from about 2 weeks ago. Packet capture shows Asterisk stops transmitting all IAX packets after this messages appears. - Dustin - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM400 FXO Screech
I just installed FXO module in an older TDM400 card in port 1 and had problems. Moved it to port 2 and everything is fine now. - Dustin - Bill Michaelson wrote: A nasty screech. That's what callers here sometimes when they dial into my FXO port from the PSTN. But usually, it works OK. Is this common? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] sill looking for a provider
The strange thing about Teliax latency is they appear to be located in Denver and it's about 55ms round trip to the Denver router on Level3 network. But then between the L3 router in Denver and the Teliax host appears to be another 30ms round trip. Which from a distance point of view makes no sense (that would be like 1500miles worth extra one way latency) . Of course it's hard for me to see the return route with traceroute. I assume the return path from their host takes on some bizarre route that adds a lot of latency. Or they are doing something strange on the final leg like a wireless link in Denver that would pump up the latency. Either way when there are multiple providers under 30ms rtt available from NYC it's a real disadvantage for Teliax to be up around 82ms. How about a nice colo datacenter guys? - Dustin - traceroute to 208.139.204.232 (208.139.204.232), 30 hops max, 38 byte packets 1 er1.nyc1.speakeasy.net (216.254.114.1) 14.236 ms 9.963 ms 9.177 ms 2 220.ge-0-1-0.cr2.nyc1.speakeasy.net (69.17.83.201) 10.721 ms 13.576 ms 7.399 ms 3 166.90.136.33 (166.90.136.33) 9.802 ms 9.790 ms 9.880 ms 4 ae-1-56.bbr2.NewYork1.Level3.net (4.68.97.161) 7.798 ms 7.202 ms 9.831 ms 5 ae-0-0.bbr1.Denver1.Level3.net (64.159.1.113) 53.055 ms 52.992 ms as-0-0.bbr2.Denver1.Level3.net (64.159.4.226) 52.191 ms 6 so-9-0.hsa2.Denver1.Level3.net (4.68.113.50) 54.296 ms so-6-0.hsa2.Denver1.Level3.net (4.68.112.162) 52.732 ms 53.188 ms 7 unknown.Level3.net (64.156.40.66) 54.259 ms 53.430 ms 52.020 ms 8 border3.ge2-0-bbnet.den.pnap.net (216.52.40.7) 54.825 ms 52.624 ms 52.293 ms 9 rockynet-1.border3.den.pnap.net (63.251.181.222) 83.526 ms 82.519 ms 83.289 ms 10 voip-co3.teliax.com (208.139.204.232) 84.333 ms 82.759 ms 82.580 ms [EMAIL PROTECTED] wrote: I tend to agree with you, my experience with Teliax has been decent, and getting better. If only I could get to them at under 20ms though, right now my latency is about 75ms whereas voipjet comes through at 19ms. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Saturday, November 05, 2005 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sill looking for a provider We have been using Teliax (www.teliax.com) for a while now and have 3 accounts with them (one for each of our asterisk servers). They've had their ups and downs but have been working to improve their support and now we are now able to speak with someone during their business hours (8-5PM MST). There was recently an issue with a high-latency link between our backbone and their backbone providers. We called them about it to find out that they had already escalated the issue with the backbone carrier and the issue was resolved. The long and short of it is that when we have had issues (which are few and far between) they have been quickly resolved. There is no question in my mind that a growing industry and growing companies are going to face customer service issues. It's bad for us now, but I think it's a good sign for the open source VoIP community. I think patience is key. Originally we looked at a number of carriers, based on the following requirements: 1) Be located in the US. 2) Have a customer support phone number and answer the phone. 3) Accept major credit cards and automatically bill my account (no need to recharge via pay-pal). 4) Allow for business/corporate usage. 5) Support IAX and g726/g729 codec's 6) Support Set Caller ID 7) Support multiple-inbound DID's We didn't care about call forwarding, voicemail, 3-way calling or any of the other features that must residential carriers tout as features. I wish I had done my research a but more formally, but the answer after about a week of research and test accounts was to use Teliax. Hope that helps. Cullin J. Wible President CEO Algorim Technologies, LLC 212-535-3238 x102 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Piotr A. Sygula Sent: Friday, November 04, 2005 6:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sill looking for a provider That concept is not bad; except when the CEO from the same company as the tech that calls all the time happens to call you from what appears to be the same caller id, and the CEO ends up hearing rap or hard rock... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Friday, November 04, 2005 5:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sill looking for a provider Jason Brashear wrote: Is there a provider that has good support and answers the phone? (= I need to get lines for my Asterisk server and want to move from broadvoice.com. So far I haven't been able to get anyone on the phone.
[Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
I have read a lot online about interrupt related cracking/popping noise issues on Digium cards. The weird thing is I experience it with the FXO port on my TDM card but not on the FXS. Does this make any sense? I had assumed an interrupt problem. But if the FXS port is working why would the FXO port be having a problem? - Dustin - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM400 FXO vs FXS Interrupt performance
I do have an older card. I believe I purchased it before the FXO module started shipping. Can I do anything to resolve or this hw problem with the card? - Dustin - Andrew Kohlsmith wrote: On Saturday 05 November 2005 22:33, Gary Eck wrote: I have popping with FSO modules only on channel 1 - the other 3 channels are clear. That was corrected a long time ago. You must have an older rev TDM400 carrier card. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] re: Attempted to delete nonexistent schedule entry...
I am having this same issue. But after I get the message all IAX processing stops. I have to reload Asterisk to get IAX peers back up and running. When I capture IAX traffic with ethereal there is no IAX messages being transmitted by Asterisk. - Dustin - On 10/15/2005, J. Iddings jeff at iddings.us http://lists.digium.com/mailman/listinfo/asterisk-users wrote: /I'm also having this issue. Everything seems to work, but it's an //unnerving error. Any thoughts? // //Jimmy wrote: // I just upgraded my test Asterisk box to the latest CVS HEAD. show // version only shows Asterisk CVS HEAD built by rootetc, with no // date or version number. I downloaded this version on Monday, Oct 3. // About once every minute, I get this while at the CLI prompt: // // sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule // entry 1! // // This only appeared after updating. All functions seem normal, other // than these messages. Phones work, auto-attendant works, voicemail works, // etc. What's going on? / OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. On 10/15/2005, J. Iddings jeff at iddings.us http://lists.digium.com/mailman/listinfo/asterisk-users wrote: /I'm also having this issue. Everything seems to work, but it's an //unnerving error. Any thoughts? // //Jimmy wrote: // I just upgraded my test Asterisk box to the latest CVS HEAD. show // version only shows Asterisk CVS HEAD built by rootetc, with no // date or version number. I downloaded this version on Monday, Oct 3. // About once every minute, I get this while at the CLI prompt: // // sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule // entry 1! // // This only appeared after updating. All functions seem normal, other // than these messages. Phones work, auto-attendant works, voicemail works, // etc. What's going on? / OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. Brett Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] determining legal VoIP service
Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. - Dustin - Walker Haddock wrote: Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier. Thanks, Walker ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] RE: Bluetooth discussions (quick glance to some BT products)
Ok I am looking at the Clipcomm solution. Bluetooth enabled phones can place voip calls by communicating with the BS-V100/L100? Am I getting this right? What is this feature called in phones and which cell phones support this? - Dustin - Eric Bart wrote: There are somes products available claiming to connect a BT headset, a cell phone and a phone land line all together. I've found some : http://www.geekzone.co.nz/content.asp?contentid=2079 http://www.clipcomm.co.kr/ The clipcomm BS-A101 sample price is : $570. It's VoIP land phone that can make/receive a voice call using your CTP cell phone or BT-enabled headset. Maybe the best system is: http://www1.norwoodsystems.com/ It's complete wireless solution that include hot connection to bluetooth hubs when the employee walks in the office. But it's not currently available. Maybe it'll work better with Bluetooth 1.2 wich will be very soon on the market. Here's a reply I got from norwoodsystems : Thanks for your enquiry. I attach a product description to help you understand our solution. We intend to sell the software through resellers and are currently reviewing the business model and packaging. We can achieve a price of under EUR450 per user including software, bluetooth headsets or phone upgrade to CTP enabled GSM phone( Bluetooth Cordless Telephony Profile), VoIP gateway, Bluetooth USB dongles, ISDN Basic Rate Interface, and software installation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Re: Digium X100P for $43
It's funny this is the second hardware counterfeiting story I have heard this week. What is going on? - Dustin - Sean Cheesman wrote: for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? -Original Message- From: Doug Meredith [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 7:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Digium X100P for $43 SamW [EMAIL PROTECTED] wrote: Digium X100P / new cards are is available on ebay for $43. Actually they seem to be made by Digit Networks. Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ADSI phone vs. IP phone
Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. - Dustin - I'm looking at ADSI phones simply because I don't have to re-tool my entire building; I can use the existing phone network and (I think) get all the functionality I need with the (far) cheaper ADSI phones. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ADSI phone vs. IP phone
I have not looked at products from every company but I do know a few offer 100mbps FastEthernet connections to the switch and the PC. - Dustin - Andrew Thompson wrote: - Original Message - From: Dustin Goodwin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 11:17 AM Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone I'm looking at ADSI phones simply because I don't have to re-tool my entire building; I can use the existing phone network and (I think) get all the functionality I need with the (far) cheaper ADSI phones. Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. - Dustin - It was my impression that these phones had 10MB ehternet connections and not 100MB. Not that most of us would notice the difference in browsing the net, it does defeat the purpose of having 100MB switches. (I believe I also saw people on this list talking about strange things happening when they wired the phones/pcs up this way.) - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] ADSI phone vs. IP phone
Seems like you have experienced a few problems implementing this type of configuration. My first hand experiences have been more positive then yours. Could you share more with the list what went wrong? Also IP phones connected to ports configured with Cisco portfast have never caused me any problems. What did you experience? - Dustin - [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Goodwin Sent: Monday, January 19, 2004 11:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. Probably because it's well known that these setups are prone to failure of either the PC's connection, the phone's connection, or degredation of one/both. It also breaks switch envirenments where spanning-tree portfast is enabled (not as big of a deal if the deployment is in concert with the infrastructure group, as it should be). Vendors should NEVER have implemented this functionality into phones unless it was working under all conditions. Personal experience shows that it is most definitely not on Cisco and 3Com products. Others have told me their stories with other manufacturer's equipment. None of it was good. It's not a production-stable way to deploy phones. Period. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Codec matching weirdness
I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. - Dustin - [EMAIL PROTECTED] wrote: I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] QoS anyone?
Rich, I would be surprised to find this. Typically ISP's will reset all QOS settings to 0 either on your CPE router if they manage it or on the aggregation router your circuit is connected to. Almost always if they support DSCP/TOS matching and priority queuing in the core of their network it's part of an extra charge service. If they don't do any priority queuing then they typical will just leave it alone and ignore it. - Dustin - Rich Adamson wrote: Snip For those that might be sending sip/iax packets across the Internet, many of the backbone ISPs now honor the QoS/TOS bit settings. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec matching weirdness
I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A So I compared the SDP info coming from the 7960, sent out from * and returning from the destination system and I have included them below. Question 1: Why is * sending out SDP info that is different from the SDP info contained in original SDP from the phone? Question 2: Is there a config option to force * to just passthrough the codec list sent by the 7960 in the invite? Question 3: What are SDP codec matching rules for SIP endpoints? How do they decide on common codec. Comparing the SDP sent and receive all systems claim support for 3 common codecs: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Now of course each device specified these 3 codecs in a different order. Under normal circumstances I feel this call should complete why is * claiming a codec mismatch? - Dustin - From phone v=0 o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68.12 s=SIP Call c=IN IP4 192.168.68.12 t=0 0 m=audio 18114 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Sent to remote server by * v=0 o=root 4205 4205 IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio 16798 RTP/AVP 4 3 0 8 2 5 10 7 18 110 97 101 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 192.246.69.223:5060 Received from remote server v=0 o=root 9755 9756 IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio 10066 RTP/AVP 18 3 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
