[asterisk-users] Polycom Provisioning Problems
Hello I am having some difficulties provisioning a set of polycom 501 phones, while another set of phones are working just fine. My Asterisk box is dual homed. On one network, where the asterisk box runs dhcpd and there are only phones, provisioning works as expected. However, for phones that are connected thru the other interface (and receive their IP address from a separate router), they are not provisioning. To add to the confusion, it seems that they fail in inconsistent ways. Even after specifying the FTP server address, name and password, these phones will complain that they cannot connect to the server, and begin loading the stored configuration. In addition, when they come up, their dates are set to Jan 1, 2001. (I think I can fix this by specifying the snmtp address, but the other phones seem to be able to find the snmtp on their own.) In inspecting the MAC-boot.log files, the phones that fail have CDP enabled, while the phones that succeed have CDP disabled. I think this is Continual Data Protection, but don't see where to disable it on the phone interface. Is this a cause of the failure? Any insight will be greatly appreciated. Thanks -- Jim Freeze ___ --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] Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation)
On Apr 20, 2007, at 7:20 PM, Matthew Rubenstein wrote: (This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. You may want to try Amazons mechanical turk. Jim ___ --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] Upgrade 4 to 8 Analog Lines Question
Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase options as follows: 1) TDM40B - use with the current TDM40B 2) Sangoma Remora A20200 - use with the current TDM40B 3) Sangoma Remora A20400 - replace the current TDM40B Any info will be greatly appreciated. Thanks Jim -- Jim Freeze ___ --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: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question
On Apr 9, 2007, at 8:32 AM, [EMAIL PROTECTED] f6hqz- [EMAIL PROTECTED] wrote: Or a new Digium TDM880B replacing the old TDM40B for only one IRQ... Do you know if this board will fit in a 2U machine? Thanks Jim Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jim Freeze Envoyé : lundi 9 avril 2007 15:15 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Upgrade 4 to 8 Analog Lines Question Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase options as follows: 1) TDM40B - use with the current TDM40B 2) Sangoma Remora A20200 - use with the current TDM40B 3) Sangoma Remora A20400 - replace the current TDM40B Any info will be greatly appreciated. Thanks Jim -- Jim Freeze ___ --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 ___ --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 ___ --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: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question
On Apr 9, 2007, at 9:29 AM, William Moore wrote: On 4/9/07, Jim Freeze [EMAIL PROTECTED] wrote: Or a new Digium TDM880B replacing the old TDM40B for only one IRQ... Do you know if this board will fit in a 2U machine? The TDM800P is about the same height as the TDM400P and is about an inch longer, so you should have no problem putting it in the same slot as the TDM400 was in. I really appreciate all the info. I think I am going to go the route of an 8 FXS card, but still need to decide between Digium and Sangoma. Are there any recommendations between: TDM880B - $679 A20400D - $906 Also, why the higher price on the Sangoma? Jim ___ --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] Adding 4 more POTS lines
Hello I have a working * server with a TDM card and 4 FXO ports. We have 4 lines now and need to add 2 more lines (and possibly two more later). I'm wondering the best upgrade path for this situation. The simplest I can invision is adding another TDM400 card with 4 FXO ports, and use 2 now and the remaining 2 later. Are there success stories with using 2 TDM cards? Any info will be appreciated. Thanks -- Jim Freeze ___ --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] Adding 4 more POTS lines
On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote: Jim, I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports, and aside from some minor echoing during peak periods, it's running smooth as ice. Hi Jay. Thanks for the info. Digium logged onto my box early on and fixed some echo problems with a code change and recompile. Do you have any theories on the cause of the echo only for peak periods? Also, I suppose there is no problem leaving 2 FXO ports unused for a time. Jim -- Jim Freeze ___ --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] Adding 4 more POTS lines
Hi Leif On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote: I have no experience with the TDM cards, but costwise it is not the best solution, in my opinion. A TDM04B (4FXO) cost around $378 at voiplink.com, while a Grandstream GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO version costs $400, almost the same as the 4FXO card. I suppose that is my alternative - remove the 4FXO card and add an 8FXO card. But I'm not seeing the prices you list. The Digium TDM2402B is listed at $837.00. Am I missing something? http://www.voiplink.com/Digium_TDM2402B_p/digium-tdm2402b.htm -- Jim Freeze ___ --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] Looking for Wireless Heaset for Polycom 501
On 10/30/06, Dean Collins [EMAIL PROTECTED] wrote: We've used the Plantronics CS50 wireless Headset with the HL10 Handset Lifter. About $240. The handset lifter leaves a lot to be desired with the 501. It lifts the handset off the cradle, but doesn't completely hang it up properly. We've had to place items under the phone to tilt it back. Other than that, the headset is great. It still amazes me that this isn't built into any handsets yetit seems totally obvious to me to put a wireless component either in or directly connected to (side card connection maybe). This handset lifter policy is just plain crap. I was wondering the same thing. But I'm still not sure if I need the lifters. I connected up the phones but didn't have time to install the lifters. The staff called today and said they have it working without the lifters. So, can someone confirm why I need the lifters? I have to agree that if the lifters are needed, then there should be a phone that comes with this built in. -- Jim Freeze ___ --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] Looking for Wireless Heaset for Polycom 501
Hi I am looking for a good wirless headset to use with the Polycom Soundpoint 501 phone. I would greatly appreciate hearing from anyone with good experiences with a specific device. Thanks -- Jim Freeze ___ --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] Looking for Wireless Heaset for Polycom 501
On 10/25/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: I've used the Plantronics ones, similar to these: http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016 and they work very well with the headset lifter, The range is pretty good too. However there are more elegant and complete solutions, with those headsets you need to be by the phone to see who is calling and to use the keypad. What are these other, more elegant complete solutions you are talking about? -- Jim Freeze ___ --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] Problems configuring Polycom 301
HiI have successfully been running with several Polycom SoundPoint 501phones and recently purchased some Polycom 301 phones.However, I can't seem to get the phones to register. The phone seesthe asterisk server, but all calls our are busy. The only difference for 'sip show peer xxx' for a working 501 phone anda non working 301 phone is:asterisk1*CLIAddr-IP : 192.168.80.204 Port 5060 # 501Addr-IP : (Unspecified) Port 0 # 301 'sip show peers' returns:asterisk1*CLI sip show peers Name/usernameHostDyn Nat ACL Port Status720/720(Unspecified)D0UNKNOWN 712/712192.168.8.205 D5060 OK (80 ms) 711/711192.168.8.203 D5060 OK (84 ms) 710/710192.168.8.204 D5060 OK (98 ms)Any 301 configuration tips would be appreciated.Thanks-- Jim Freeze ___ --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] Prevent a Polycom contact list to be overwritten
On 8/3/06, Stephen Murphy [EMAIL PROTECTED] wrote: I have a Polycom phone that was setup without provisioning through an FTP server. It has a number of contacts that where input via the phone. I would like to add this phone to a small network that was provisioned through an FTP server and keep the contacts already on the phone. How do I ensure that the contacts list file will not be overwritten when I do a provisioning?I would like to know this as well, but for a slightly different reason. I want to provision501 phones, but I want to start from what is currently on the phone. So, in other words, I want to download the XML file that is stored in the phone.Anyone know how to do this?Thanks-- Jim Freeze ___ --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] Polycom Soundpoint IP 301 w/ MGCP
HiI was about to order a polycom 301 when I noticed that the VoIP protocol is listedas MGCP and not SIP, as with the 501.First, what is MGCP?And, will the 301's work seamlessly with Asterisk and the other 501 phones that I have?Thanks Jim Freeze ___ --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] ExternalIVR vs AGI
On Jun 27, 2006, at 10:19 AM, Daniel Salama wrote: I have an Perl AGI script that acts as an IVR for my Asterisk box. Basically, it simply plays audio files to the caller, collecting DTMF input and logging every DTMF input into a database table, simply to document every step or option selected by the caller. One thing is that in addition to playing audio files, it also, at some point, plays SayUnitTime command. This IVR has constantly about 20 simultaneous callers 24x7. Would it be more resource efficient to migrate this to ExternalIVR? What are the pros/cons of using ExternalIVR vs using my Perl AGI. Probably no difference. The ExternalIVR lets you interrupt actions (like playing a recording) where the AGI will not. But you don't need that. If the caller hangs up early (say, in the middle of a playback), I think both AGI and ExternalIVR quit immediately. Jim Thanks, Daniel ___ --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 Jim Freeze ___ --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] Weird Can-Reinvite problem
Have you tried turning off icmp redirect on your router?On 6/6/06, Brett N [EMAIL PROTECTED] wrote: Hi All,I'm having a really weird can reinvite issue. I've been banging my head around on this for days now..I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5172.20.0.11 is a hosted box and serves multiple offices172.20.2.5 is a box on site at a customer's office.A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phoneat 172.20.2.80 via server 172.20.2.5:Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to eachother. so phone a can ping phone b and server b, etc, etc, etc..Can reinvite is enabled on both the ser connection (on both sides) and for both phones..Making a call from phone A to phone B works great.. Except you can hear apop when the reinvite happens. After the call is connected Phone B cantransfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another localextension on asterisk A) the call will have 0 way audio. If the call istransfered back, there will be one way audio.It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THENthe originator can transfer..I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinviteslooks ok..No Nat, no funny business here.. just IP routing.. Any ideas?-Brett___--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 -- Jim Freeze ___ --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] Running down an echo problem on outgoing calls
HiI am having some echo on outgoing calls on a pstn line (made from a sip phone)and am looking into some fixes involving rxgain and txgain, butbefore I go down that road, I want to make sure I have everythingsetup correctly.My zapata.conf has cat /etc/asterisk/zapata.conf [trunkgroups]; define trunk groups[channels]; hardware channels; defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechocancelwhenbridged=yesechotraining=yessignalling=fxs_ks; define channelscontext=pstn-incomingchannel = 1-4But, when I run asterisk and ask it about the zap channels, I get that echo cancellationis OFF.zap show channel 2Echo Cancellation: 128 taps, currently OFFCan someone please explain this?Thanks Jim Freeze ___ --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] Volume configuration on Polycom Soundpoint 501 phone
We are using the polycom 501 phones, and are having some challengeswith the volume setting. When a phone call comes in, the user ups thevolume for the handset, but they have to repeat that for every call.Currently, the volume level seems to reset itself at about 60%. Is there a way for the user to change their default volume level?Thanks-- Jim Freeze ___ --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 help with getting EXTEN from pstn hunt group
Hi
On Apr 22, 2006, at 9:02 PM, Leo Ann Boon wrote:
An example:
- assuming a hunting pstn number 2341000
- 4 lines in a the group: 2341001, 2341002, 2341003, 2341004
- The 4 lines, are connected to your TDM card as Zap/1, Zap/2, Zap/
3, Zap/4
If you want to find out which line was actually called, put each
line into a different context in your zapata.conf, e.g.
context=pstn_2341001
channel=1
context=pstn_2341002
channel=2
...
In your extension.conf, you'll need something like
[psnt_2341001]
exten = s,1,Answer()
exten =s,2,Set(DNID,2341001)
All seems to work up to this point, But I am having problems with the
Set() command.
I think the Set() syntax is
Set(DND=2341001)
but when I do SayAlpha(${DNID}) or SayDigits(${DNID}), the output is
always and
empty string.
exten =s,3,Goto(defuault,s,1) ; Jump to normal processing
[psnt_2341002]
exten = s,1,Answer()
exten =s,2,Set(DNID,2341002)
exten =s,3,Goto(defuault,s,1); Jump to normal processing
Hope this helps.
Yes, much. Thanks
--
Jim Freeze
___
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Re: [Asterisk-Users] Need help with getting EXTEN from pstn hunt group
On Apr 22, 2006, at 9:02 PM, Leo Ann Boon wrote: Jim, You might want to be a little more specific: a. You want to find out which line the call came in on, OR b. The actual PSTN number that was dialed As I think about this more, and considering the solution you provide, I think there is still a problem. Let me re-explain. I have four numbers on a hunt group: Zap/1-1 234-1000 Zap/2-1 567-1002 Zap/3-1 567-1003 Zap/4-1 567-1004 The published number that everyone calls is 234-1000. I want all calls to that number to be sent to the front desk. That means, that even if the 1000 line is busy, and the next open line found is 567-1002, I still wan the call to go to the front desk. But, if someone calls 1002-1004, I want those calls to be directed to various phones in the office. So, it seems that it is not sufficient to know what zap channel the call comes in on, but we need the number dialed by the caller. Is this possible to do? Thanks -- Jim Freeze ___ --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] Need help with getting EXTEN from pstn hunt group
Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten = s,1,Answer() exten = s,n,Dial(ZapZap) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten = 1234567,1,Answer() # never gets here I thought I could get the extension on the 's' extensions above, but, no, the extension is 's'. Is there something special that needs to be done with pstn hunt groups to get the extension? Thanks -- Jim Freeze ___ --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] FXS or VOIP
Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. Thanks -- Jim Freeze ___ --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] FXS or VOIP
On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: You can save a little money with analog phones however if that saving is not an issue business class VoIP phones from providers like Polycom and Cisco have more features and much of the time better call quality. Thanks William for the response. That is good news about the phone quality. From what I have read, I think the overall cost would still be cheaper with a voip solution, even if the phones are more. A 4 line FXS card is about $3-400 (I think). If I understand this correctly, even if I have only 4 lines incoming, I need an FXS homerun to each phone. So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. I was hoping a single computer could handle up to 10 voip phones. Am I deluding myself? Jim Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. -- Jim Freeze ___ --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] FXS or VOIP
Hi Colin On 1/11/06, Colin Anderson [EMAIL PROTECTED] wrote: On the upside: [snipped] On the downside: [snipped] hth Yes. Thanks. That helped a lot. You certainly make FXS sound scary (black art). For a new office setup, it seems like the better choice is voip all the way. -- Jim Freeze ___ --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] FXS or VOIP
On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: A single computer will handle hundreds of telephones. Just get a card with more ports, or use an external gateway. I am sorry, I don't understand. Are you talking about analog FXS phones? All the PCI cards I have seen have a max of 4 FXS lines and the external boxes seem very expensive. -- Jim Freeze ___ --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] FXS or VOIP
Hi Bryan On 1/11/06, Copeland, Bryan [EMAIL PROTECTED] wrote: In one of my configurations I have an office with a digium dual T1 card... 23 Channel PRI coming in on port 1... Coastcom mux on port2 breaking out FXS channels.. and voip phones... Initially the plan was to use voip phones and use FXS channels for fax machines.. After putting some phones on the FXS channels and some on the voip.. I would choose FXS as my primary solution.. The reliability and call quality was better in our solution.. I will be doing another installation with only FXS channels and ADSI phones... Wow, that sounds similar to my planned setup. Why do you suppose that the FXS lines were better than voip in your solution. Your results seem opposite of what the majority have been singing. Were your voip phones sharing a network with data traffic? Were the voip phones of a good quality? Did you have problems tweaking the FXS system? Thanks -- Jim Freeze ___ --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
