[Asterisk-Users] (Un)PauseQeueMamber usage
Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe ___ --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] Simple question about ringing multiple phones (extensions)?
Hey Gurus, I have a very simple asterisk setup that basically lets me share a PSTN line from one location to another. I would like to have the phones at both locations ring when the PSTN # is dialed(inbound calls from PSTN to asterisk). I tried something like: exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr) I thought this might cause both 2005 and 2010 to ring when 2020 was dialed, but only 2005 rings? Thanks for ideas or suggestions on this. Marty ___ --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] No IN and OUT on ISDN line at the same
Hello Armin, The card is telling: CAPI INFO 0x34a2: No circuit / channel available so the other channel must be in use by something else. Maybe another device on the ISDN line? I have tested it several times now and always entered capi info before and after the call. The answer was always: Contr1: 2 B channels total, 2 B channels free. I'm currently alone in the office, no incoming/outgoing faxes, no incoming/outgoing calls. Is there a chance for me to figure out who or what is using the other B channel while the call is coming in? Thanks for your help, Ralf -- ___ Play 100s of games for FREE! http://games.mail.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
RE : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports
Buy a TDM2400P card with several quadFXO modules : 24 ports max :-) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de roswel ajf Envoyé : vendredi 27 janvier 2006 23:17 À : asterisk-users@lists.digium.com Objet : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel version. That code is working only with 8 or less ports (accumulative) on digium fxs/fxo cards (2 cards with 4 ports each). the questoin is, what if we want 12 ports?..well, really, i don't understand the limitations? is it simply zaptel driver code fix? or kernel fix? or technology limitation? donno any tips would help. we are though planning to move to latest asterisk 1.2.3 on linux 2.4. thanks, very much appreciate any comments. ___ --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: [Asterisk-Users] Simple question about ringing multiple phones (extensions)?
Martin Joseph wrote:
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a
PSTN line from one location to another. I would like to have the
phones at both locations ring when the PSTN # is dialed(inbound calls
from PSTN to asterisk).
I tried something like:
exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL=${PHONE_601}${PHONE_602}${PHONE_603}
PHONE_601=SIP/601; office 601 Ronald
PHONE_602=SIP/602; office 602 Ronald
PHONE_603=ZAP/1r1; living room 603 cordless
For you this should work too:
exten = 2020,2,Dial(SIP/2005IAX/2010,25,tr)
bye
Ronald Wiplinger
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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] No IN and OUT on ISDN line at the same
On Sat, 28 Jan 2006, Ralf Mueller wrote: Hello Armin, The card is telling: CAPI INFO 0x34a2: No circuit / channel available so the other channel must be in use by something else. Maybe another device on the ISDN line? I have tested it several times now and always entered capi info before and after the call. The answer was always: Contr1: 2 B channels total, 2 B channels free. Okay, that means that Asterisk/chan_capi isn't using a channel at that time. But it does not know about other programs or even other devices on the ISDN bus. When the call is coming in, are you sure you don't try to forward it to more than one CAPI destinations? For each destination, one channel is needed, even if the call is not accepted. I'm currently alone in the office, no incoming/outgoing faxes, no incoming/outgoing calls. Is there a chance for me to figure out who or what is using the other B channel while the call is coming in? A dchannel trace might show something. Armin ___ --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] Simple question about ringing multiple phones(extensions)?
Marty, Just remove the options for each technology. Dial(SIP/2005IAX/2010,25,tr) This should do the job Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: zaterdag 28 januari 2006 9:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Simple question about ringing multiple phones(extensions)? Hey Gurus, I have a very simple asterisk setup that basically lets me share a PSTN line from one location to another. I would like to have the phones at both locations ring when the PSTN # is dialed(inbound calls from PSTN to asterisk). I tried something like: exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr) I thought this might cause both 2005 and 2010 to ring when 2020 was dialed, but only 2005 rings? Thanks for ideas or suggestions on this. Marty ___ --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
[Asterisk-Users] Trunk is not released
Hi! I have this little problem here and i really don't know how to solve it. This is the scenario: I've setup a IVR, using my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However, after my mobile hang up, the Trunk Zap1 does not. I've to reboot the computer to free up the line and it is also not possible to do a graceful reboot because I would get a kernel panic. I'm actually using PSTN for the trunk. Hope anyone can provide me some advice, couldalso be a linkto another post which I might have missed when searching for my answers. Thx inadvance Jeremy ___ --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] Lockups since upgrade 1.2.3 - anyone else? Anyideas?
Same here on a fresh install on a test machine, no TDM card, just fresh install using fresh install of CentOS-4.2.ServerCD-i386.iso. I thought it was more likely when we accessed Flash Operator Panel. Want to upgrade from 1.0.9 but now a bit more worried. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Saturday, 28 January 2006 5:42 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Anyideas? Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for business. I have scoured the logs, and find no errors. Not even right before/around the time of the crash. I am worried that 1.2.3 is not as stable as 1.0.9 (or 1.0.10, though we never ran that version). Is there a needed step aside from make; make install that I missed when upgrading? Has anyone else had similar problems? Or, if I submit other info, would someone have a clue as to what to look at? We run a TDM400P with 3 FXO modules, and about 15 SIP Cisco 79XX phones here. Any help is appreciated, this cannot continue. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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 ___ --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] No IN and OUT on ISDN line at the same
This could be a context issue, I had to fuss with mine to get the channels working independently too. I'll try to post the examples tomorrow, way to tired now :). Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: Saturday, January 28, 2006 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same On Sat, 28 Jan 2006, Ralf Mueller wrote: Hello Armin, The card is telling: CAPI INFO 0x34a2: No circuit / channel available so the other channel must be in use by something else. Maybe another device on the ISDN line? I have tested it several times now and always entered capi info before and after the call. The answer was always: Contr1: 2 B channels total, 2 B channels free. Okay, that means that Asterisk/chan_capi isn't using a channel at that time. But it does not know about other programs or even other devices on the ISDN bus. When the call is coming in, are you sure you don't try to forward it to more than one CAPI destinations? For each destination, one channel is needed, even if the call is not accepted. I'm currently alone in the office, no incoming/outgoing faxes, no incoming/outgoing calls. Is there a chance for me to figure out who or what is using the other B channel while the call is coming in? A dchannel trace might show something. Armin ___ --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: [Asterisk-Users] Trunk is not released
you need to get your zap line to listen for the hang-up. in my /etc/asterisk/zapata.conf: busydetect=yes busycount=3 make sure you have the right loadzone set up in /etc/zaptel.conf =) [EMAIL PROTECTED] wrote: Hi! I have this little problem here and i really don't know how to solve it. This is the scenario: I've setup a IVR, using my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However, after my mobile hang up, the Trunk Zap1 does not. I've to reboot the computer to free up the line and it is also not possible to do a graceful reboot because I would get a kernel panic. I'm actually using PSTN for the trunk. Hope anyone can provide me some advice, couldalso be a linkto another post which I might have missed when searching for my answers. Thx inadvance Jeremy ___ --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: [Asterisk-Users] Nagios and Asterisk
On 09:20, Sat 28 Jan 06, James Harper wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Darrell Long Sent: Saturday, 28 January 2006 05:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nagios and Asterisk Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. I'm not doing that but dropping a call file in should do the trick shouldn't it? Along the same lines, does anyone know of any snpp servers that are compatible with app_sms? I have nagios on another server and would like to send pages via app_sms and so an snpp server running on the asterisk server would be a good way to go about it. We stopped using asterisk for it and switched to bayham sms :) They provide a perl agi that is easy to change so it doesn't need asterisk to send sms :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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: [Asterisk-Users] Trunk is not released
the 3 is the number of busy tones it listens for. With Australian signalling, there is a 'high low' sound, 'beep beep' on hang-ups, once it hears three of these, it will terminate the line. - in my set-up, it should hang up with in 5 seconds of the hang-up. im not too sure about the problem you've decribed though. as Asterisk appears to have already hung up theline (as indicated from FOP) if you are on the console, and type: soft hangup (tab) does it list the zap channel still? [EMAIL PROTECTED] wrote: thx :) yup i've done that. but it was at busycount=6. anyway i've changed to 3 but it still takes more than 5 mins to clear the trunk. I couldn't wait till it clears on its own, so i can't give an exact time. I've also noticed that at the FOP, the trunk has already turned into GREEN, however, when trying to call in, it still shows the line is busy, so I'm wondering if this problem is actually related to the zapata.conf. Since all other scenarios of "hang-ups" I've tested so far doesn't seem to be giving a problem with the trunk. How does the 3 relates to the busy tone, I mean like How many mins does it take for 1 busy tone to be produced ? - Original Message - From: Tom Paseka To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, January 28, 2006 6:00 PM Subject: Re: [Asterisk-Users] Trunk is not released you need to get your zap line to listen for the hang-up. in my /etc/asterisk/zapata.conf: busydetect=yes busycount=3 make sure you have the right loadzone set up in /etc/zaptel.conf =) [EMAIL PROTECTED] wrote: Hi! I have this little problem here and i really don't know how to solve it. This is the scenario: I've setup a IVR, using my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However, after my mobile hang up, the Trunk Zap1 does not. I've to reboot the computer to free up the line and it is also not possible to do a graceful reboot because I would get a kernel panic. I'm actually using PSTN for the trunk. Hope anyone can provide me some advice, couldalso be a linkto another post which I might have missed when searching for my answers. Thx inadvance Jeremy ___ --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: [Asterisk-Users] DTMF's indescipherable, but voice clean!
Ah yes, quite relevant details. It is a VoIP SIP-based DID. This problem is so strange because it suddenly started happening. What other info is relevant?GabeOn 1/27/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn-pbx is unsuccessful, while pbx- pstn works fine. The tones simply don't make it through. Tiny brief fragments are all.It might help to describe what interfaces your Asterisk PBX with the PSTN. Is this a VoIP provider DID you are using, or a POTS line with an interfacecard, or a PRI with a digital interface card?Nabeel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] double ringing tone on asterisk 1.2 (workaround)
After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem changing the call progress tone of sip phones to something not udible. ___ --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] G729 Commercial Licenses.
Rob, Thanks, well i had gone through it before but i had some different comments from a couple of friends on the same topic but let me clarify. currently i have 2 commercial licenses and suppose i a have backups of the licenses and once a i do a full revamp and i place my .lic files back at the respective folders... im gonna have a sure go on the same PC? am i right..? Please correct me if i am wrong. Thanks Dan On 28/01/06, Rob Lith [EMAIL PROTECTED] wrote: Read towards the bottom of http://www.digium.com/downloads/ftp/asterisk/g729/READMERob On 1/28/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my [EMAIL PROTECTED].Just want to clarify some licensing issues regarding them. If i had to do a full reformat of my PC and reload [EMAIL PROTECTED] again will i be able to use the licenses again without re-registration? If no. .Is there are limits for this? Please anyone clarify. Thanks Dan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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: [Asterisk-Users] (Un)PauseQeueMamber usage
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Does anyone have an example of hoe to use these two commands? I have read he
documentation, and I am still unclear on where this command goes, as part of
extensions.conf or where?
If someone could post a working example it would be most helpful.
Here's how I've done it before for other clients:
On the dialout portion I've changed the dial plan to:
exten = _1NXXNXX,1,GotoIf($[${LEN(${$[AGENTBYCALLERID_${CALLERID(num)}]})}
2]?2:3)
exten =
_1NXXNXX,2,PauseQueueMember(|Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]})
exten = _1NXXNXX,3,Dial(SIP/SIP PEER/${EXTEN},,Tg)
exten = _1NXXNXX,4,ForkCDR()
What that's basically saying is that if the calling number is also
logged in as an agent, go ahead and pause that queue member in all
queues that they belong to and then make the call. I'm doing the
GotoIf because there are other extensions in that same context that
may not be logged in as agents and I don't want to make that pqm call
(though there's no real harm in doing so, it'll just tell you there's
no Interface as specified) with.
Then, in that same context, you put the following in the h extension
exten = h,1,ForkCDR()
exten = h,2,GotoIf($[${LEN(${$[AGENTBYCALLERID_${CALLERID(num)}]})} 2]?3:4)
exten = h,3,UnPauseQueueMember(|Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]})
exten = h,4,NoOp(Done!)
ForkCDR is important because if you don't do it you're going to find
that the original CDR that used to contain the destination number in
it, now contains only the 'h' extension in it. You could also use
ResetCDR(w) here. Your choice really. ForkCDR will fork the one CDR
into two preserving the original dial information, and then you may
choose to do a NoCDR() or just deal with the additional CDR generated
to the 'h' extension by ignoring it when you parse CDRs.
Hope this helps.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
I have different need. In the same issue Vic presents. It's 3000 concurrent calls from PSTN (E1s) to Voip (gsm). And the other way around. 3000 Voip calls (SIP/H323 gsm) to PSTN. no voicemail, but the user may get 5 seconds of help prompts initially. Thanks, On 1/28/06, Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] OT?: International number parsing
Ron hotmail ha scritto: The short answer is no, you will never have a situation where the 'local' part of the term number is mistaken for part of the dialcode. for example, your customer dials 0119647701773352 (Iraq mobile number) Iraq011964 Iraq-Baghdad 0119641 Iraq-Mobile 0119647701 this would cause a match on Iraq, and Iraq-Mobile, but not on baghdad, the 'most' accurate match would be the dialcode with the most digits... That's the way I'm doing it : let's MAX_PRE_LENGHT be the maximum lenght for a prefix (as today it's 10, for 0061891006, Australia Christmas Island) and DST_LENGTH the lenght of the called number (DST) for i in range(min(MAX_PRE_LENGTH, DST_LENGTH)): probablePrefix = DST[0:min(MAX_PRE_LENGTH, DST_LENGTH)-i] select probablePrefix from a table with all the prefixes (and other info you can need) if we found something that's the prefix, break to the application else continue with a smaller try From the original post it seems there are two tables, one for the country and one for the city, like having one table with 0011964 - Iraq and one with Iraq - 1 - Bagdad Iraq - 7701 - Mobile I don't know if this speed up things, in my case it surely won't since I have a large-grained detail for locating the call (I'm not interested in city codes, so for example I've only one entry for Italy, and not a lot of entries like 'Italy Milan', 'Italy Rome'...) so a join would slow the benefit of smaller values for MAX_PRE_LENGTH, it depends on the application. Seems that when you need to have fine-grained detail the search is made in reverse, for example message boxes for cellular phones : black box understanding warning if I call (not a real number, but I know a real example I won't post for obvious reasons) 345 - 333444555 while the cell is off I get a voice : answer 333444555, the phone is off, leave a message if I call 345 - 333444555 the message is the same : answer 333444555, the phone is off, leave a message so the search is made backwards, and the application starts as long as only one possible match is found. I don't even think we are talking about relational db here, probably some directory to speed up things with a tree-search, anyone working in the large who can confirm ? /black box understanding warning ___ --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] G729 Commercial Licenses.
[EMAIL PROTECTED] wrote: Rob, Thanks, well i had gone through it before but i had some different comments from a couple of friends on the same topic but let me clarify. currently i have 2 commercial licenses and suppose i a have backups of the licenses and once a i do a full revamp and i place my .lic files back at the respective folders... im gonna have a sure go on the same PC? am i right..? Please correct me if i am wrong. According to the license, it's based on MAC address, as long as that doesn't change you should be all set: A G.729 key must be re-registered if any of the ethernet devices in your Asterisk server are changed, added, or removed. The unique G.729 license file which is located in your /var/lib/asterisk/licenses directory is tied to the MAC address of all the ethernet devices installed in your system. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] regarding connecting to AMP
hi alli have intalled [EMAIL PROTECTED] successfully and now the problem is that how can i connect to AMP so that i would be able to configure it.actually i have following setup... one [EMAIL PROTECTED] machine and two other machines i want that these two clients machine can be able to call each other through using [EMAIL PROTECTED] box.i connect this [EMAIL PROTECTED] boxto the hub...(simple hub)...now tell me what ip scheme i would use to configure it ...and how it would be possible to complete my task...one more thing i have also xlite sip phone ...i will call these two machine through these sip soft phonesnow plz temme complete idea becaz i have no good experience about it.i shall be thankful to you BYE-- Muhammad Sohail ArhamU.E.T. LahorePhone No. 0321-4422406 ___ --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] G729 Commercial Licenses.
Thanks. Doug for the precise clarifications..DanOn 28/01/06, Doug Lytle [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Rob, Thanks, well i had gone through it before but i had some different comments from a couple of friends on the same topic but let me clarify. currently i have 2 commercial licenses and suppose i a have backups of the licenses and once a i do a full revamp and i place my .lic files back at the respective folders... im gonna have a sure go on the same PC? am i right..? Please correct me if i am wrong. According to the license, it's based on MAC address, as long as thatdoesn't change you should be all set:A G.729 key must be re-registered if any of the ethernet devices in your Asteriskserver are changed, added, or removed.The unique G.729 license file which islocated in your /var/lib/asterisk/licenses directory is tied to the MAC address ofall the ethernet devices installed in your system.Doug--Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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: [Asterisk-Users] Skype-to-Asterisk(SIP): progress
However, for a small sub-set of users that I work with, Skype is a channel that is preferred for audio in some circumstances, and I feel that it's worthwhile to have some ability to connect with users who have expressed that preference. Thanks for your post, John. I too encounter resistance when I ask subcontractors in other countries to use X-Ten or other clients to connect to our pbx. The invariable, ya, I use Skype doesn't inspire me, since I'd have to be a a computer to use it too. You almost have to mail them a hardware phone to get them to do it. (Then they're hooked by the way.) Cellphones have lessened the importance of being able to reach someone in the same city by using their pbx directly (alas, they'll always find you if your cell is on), but the story is still the same for people in different parts of the world. It'd be great if * could talk to Skype, especially natively. Maybe someday it will be an advantage to Skype but right now it's like you say - competition. ___ --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] Name/username (sip show peers)
On Sat, 2006-01-28 at 13:13 +0800, Ronald Wiplinger wrote:
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
I'm not entirely sure what you want here, but something like this might
be easier to read:
[globals]
hotline=SIP/601
[local-sip]
exten = 601,Dial(${hotline},30,r)
HTH
Pete
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RE : [Asterisk-Users] G729 Commercial Licenses.
Title: Message I have 25 licences here, u will have the possibility to Re-register once in case of failure, even if your mac-addresses are different. After that, they will ask you some explanations. Olivier -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED]Envoyé: samedi 28 janvier 2006 14:36À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: Re: [Asterisk-Users] G729 Commercial Licenses.Thanks. Doug for the precise clarifications..Dan On 28/01/06, Doug Lytle [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Rob, Thanks, well i had gone through it before but i had some different comments from a couple of friends on the same topic but let me clarify. currently i have 2 commercial licenses and suppose i a have backups of the licenses and once a i do a full revamp and i place my .lic files back at the respective folders... im gonna have a sure go on the same PC? am i right..? Please correct me if i am wrong. According to the license, it's based on MAC address, as long as thatdoesn't change you should be all set:A G.729 key must be re-registered if any of the ethernet devices in your Asteriskserver are changed, added, or removed.The unique G.729 license file which islocated in your /var/lib/asterisk/licenses directory is tied to the MAC address ofall the ethernet devices installed in your system.Doug--Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Anyideas?
Hi James - Same here on a fresh install on a test machine, no TDM card, just fresh install using fresh install of CentOS-4.2.ServerCD-i386.iso. I thought it was more likely when we accessed Flash Operator Panel. Want to upgrade from 1.0.9 but now a bit more worried. Does anything get recorded in either the system or asterisk logs prior to the lock-up? - Noah ___ --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] test
test ___ --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] G729 Commercial Licenses.
Title: Message test From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Taylor Sent: Saturday, January 28, 2006 8:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] G729 Commercial Licenses. I have 25 licences here, u will have the possibility to Re-register once in case of failure, even if your mac-addresses are different. After that, they will ask you some explanations. Olivier -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé: samedi 28 janvier 2006 14:36 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [Asterisk-Users] G729 Commercial Licenses. Thanks. Doug for the precise clarifications.. Dan On 28/01/06, Doug Lytle [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Rob, Thanks, well i had gone through it before but i had some different comments from a couple of friends on the same topic but let me clarify. currently i have 2 commercial licenses and suppose i a have backups of the licenses and once a i do a full revamp and i place my .lic files back at the respective folders... im gonna have a sure go on the same PC? am i right..? Please correct me if i am wrong. According to the license, it's based on MAC address, as long as that doesn't change you should be all set: A G.729 key must be re-registered if any of the ethernet devices in your Asterisk server are changed, added, or removed.The unique G.729 license file which is located in your /var/lib/asterisk/licenses directory is tied to the MAC address of all the ethernet devices installed in your system. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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: [Asterisk-Users] Name/username (sip show peers)
Pete Barnwell wrote:
On Sat, 2006-01-28 at 13:13 +0800, Ronald Wiplinger wrote:
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
I'm not entirely sure what you want here, but something like this might
be easier to read:
[globals]
hotline=SIP/601
[local-sip]
exten = 601,Dial(${hotline},30,r)
That still will give you with sip show peers (as mentioned in the
Subjectline) still:
601/601
But I would like something:
hotline/601or601/hotline
123456789/Peter_Pan or Peter_Pan/123456789
.
bye
Ronald Wiplinger
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[Asterisk-Users] Voip Provider
Hi Everyone, I know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100s to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service. I use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. Ive been looking for service for 2 months and I havent had any luck. P.S. I do not need any special services, just proper DTMF tone handling. Mark Adams Infinity Marketing 1-800-430-1478 Main 530-579-8856 Fax ___ --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] Voip Provider
Title: Message Hi, feel free to contact me off-list, we can have a test if you want. [EMAIL PROTECTED] -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mark AdamsEnvoyé: samedi 28 janvier 2006 15:50À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] Voip Provider Hi Everyone, I know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100s to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service. I use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. Ive been looking for service for 2 months and I havent had any luck. P.S. I do not need any special services, just proper DTMF tone handling. Mark AdamsInfinity Marketing 1-800-430-1478 Main 530-579-8856 Fax ___ --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: 5, 000 concurrent calls system rollout question
What about IAX - SIP or IAX - IAX? Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 28, 2006 5:43 AM Subject: Asterisk-Users Digest, Vol 18, Issue 185 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Context for SIP incoming (newbie question?) (Nabeel Jafferali) 2. RE: DTMF's indescipherable, but voice clean! (Nabeel Jafferali) 3. Re: Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2? (Rob Lith) 4. Re: G729 Commercial Licenses. (Rob Lith) 5. Re: 5,000 concurrent calls system rollout question (Rob Lith) 6. Re: 5,000 concurrent calls system rollout question (Leo Ann Boon) 7. (Un)PauseQeueMamber usage (Joe) 8. Simple question about ringing multiple phones (extensions)? (Martin Joseph) 9. Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same (Ralf Mueller) 10. RE : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports ([EMAIL PROTECTED]) 11. Re: Simple question about ringing multiple phones (extensions)? (Ronald Wiplinger) 12. Re: 5,000 concurrent calls system rollout question (Zoa) 13. Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same (Armin Schindler) 14. RE: Simple question about ringing multiple phones(extensions)? (Henk Dick) 15. Trunk is not released ([EMAIL PROTECTED]) 16. RE: Lockups since upgrade 1.2.3 - anyone else? Anyideas? (James Sturges) 17. RE: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same ([EMAIL PROTECTED]) 18. Re: Trunk is not released (Tom Paseka) 19. Re: Nagios and Asterisk (Michiel van Baak) 20. Re: shared fxo line (Wilson Pickett) 21. Re: Trunk is not released (Tom Paseka) -- Message: 12 Date: Sat, 28 Jan 2006 11:03:56 +0200 From: Zoa [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-2022-JP It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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: [Asterisk-Users] Re: Polycom 501 horrible echo
One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). That has got to be the problem! I'll let you know how the results go. Thanks ___ --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] sip registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ 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
[Asterisk-Users] sip registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ 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
Re: [Asterisk-Users] sip qualify=yes interval
So; qualify=1000|yes means query for SIP OPTIONS, then take then unregister the peer if no response in 1000ms. But, how do you set/determine the frequency at which a peer is queried? Does this go on indefinitely after a peer fails to respond to make sure the peer is re-registered when available again? Can the interval be set on a per peer basis? There are no hard and fast rules in terms of what value is used. It all depends 100% on the reliability of your sip connections, and what might be good for me may not even come close to addressing your needs. In other words, the more reliable your sip connections are (end-to-end), the greater the value can be. I've got multiple remote sip phones where reliability is usually not an issue, and setting qualify=1 (ten seconds) is fine. The trade-off is the lower the value, the more sip traffic generated. If your asterisk box is behind a low speed dsl connection or on a broadband connection that gets charged for usage exceeding a certain traffic volume limit, you might want to use a larger qualify value. If bandwidth is not an issue and reliability is a major issue, then use a low value. It is sort of like setting rx and tx gains on analog pstn cards; there is no such thing as a standard value that works for everyone. ___ --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] No IN and OUT on ISDN line at the same
Ok my examples are here for capi: Simple but works. http://www.voip-info.org/wiki/view/Example+North+American+CAPI+Setup Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Saturday, January 28, 2006 4:53 AM To: asterisk-users@lists.digium.com Subject: RE: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same This could be a context issue, I had to fuss with mine to get the channels working independently too. I'll try to post the examples tomorrow, way to tired now :). Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: Saturday, January 28, 2006 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same On Sat, 28 Jan 2006, Ralf Mueller wrote: Hello Armin, The card is telling: CAPI INFO 0x34a2: No circuit / channel available so the other channel must be in use by something else. Maybe another device on the ISDN line? I have tested it several times now and always entered capi info before and after the call. The answer was always: Contr1: 2 B channels total, 2 B channels free. Okay, that means that Asterisk/chan_capi isn't using a channel at that time. But it does not know about other programs or even other devices on the ISDN bus. When the call is coming in, are you sure you don't try to forward it to more than one CAPI destinations? For each destination, one channel is needed, even if the call is not accepted. I'm currently alone in the office, no incoming/outgoing faxes, no incoming/outgoing calls. Is there a chance for me to figure out who or what is using the other B channel while the call is coming in? A dchannel trace might show something. Armin ___ --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: [Asterisk-Users] Name/username (sip show peers) - ok in sip.conf, but how in REAL-TIME?
Pete Barnwell wrote: On Sat, 2006-01-28 at 22:32 +0800, Ronald Wiplinger wrote: [snip] asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status PeteB/peteb(Unspecified)D 0UNKNOWN The username is the username the sip device registered with, the name is the start of the definition (not sure what the official name for it is!) This is the def in sip.conf:- [PeteB] type=friend careinvite=no qualify=yes username=peteb secret=123456 host=dynamic context=internal dtmfmode=rfc2833 Think that covers it ;) Thanks Pete, this works now well if it is in sip.conf, but I had no success with it if I use it in Real-time. What do I need to do there? bye Ronald Wiplinger ___ --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] DTMF's indescipherable, but voice clean!
OK Computer wrote: Ah yes, quite relevant details. It is a VoIP SIP-based DID. This problem is so strange because it suddenly started happening. What other info is relevant? Maybe the company providing the DID so other people can say if they are having problems, and the SIP debug on a call that has problems. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Other side disconects when using TxFAX
I'm want to send a fax, but its failing with Unicall/XX event Far end
disconnected , right after the Answer command. Any tips?
TIA,
--
Paulo
[extensions.conf]8-
[txfax]
exten = s,1,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,NoOp(${FAXFILE})
exten = s,n,Playback(pleaseineed2faxyou)
exten = s,n,TxFAX(${FAXFILE}|caller|debug)
exten = t,2,Hangup
[extensions.conf]8-
[messages]8-
-- Attempting call on UniCall/30/33485018 for [EMAIL PROTECTED]:1 (Retry 1)
Jan 28 14:10:07 WARNING[18949]: chan_unicall.c:2644 handle_uc_event:
Unicall/30 event Dialing
Jan 28 14:10:11 WARNING[18949]: chan_unicall.c:2644 handle_uc_event:
Unicall/30 event Alerting
Jan 28 14:10:16 WARNING[18949]: chan_unicall.c:2644 handle_uc_event:
Unicall/30 event Connected
Channel UniCall/30-1 was answered.
-- Executing Set(UniCall/30-1, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
-- Executing Set(UniCall/30-1, TIMEOUT(response)=10) in new stack
-- Response timeout set to 10
-- Executing Answer(UniCall/30-1, ) in new stack
-- Executing Wait(UniCall/30-1, 1) in new stack
Jan 28 14:10:17 WARNING[18949]: chan_unicall.c:2644 handle_uc_event:
Unicall/30 event Far end disconnected
Jan 28 14:10:17 WARNING[18949]: chan_unicall.c:2930 handle_uc_event: CRN
32832 - far disconnected cause=Normal Clearing [16]
-- Channel 0 got hangup
[messages]8-
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[Asterisk-Users] VOIP carriers and asterisk
Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? thanks Burak Balasaygun ___ --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] Polycom 501 horrible echo
BootBlock 2.5.0 Bootrom 2.6.2.0032 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Senykoff Sent: January 27, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 horrible echo I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Which bootrom version are you using? -Ron ___ --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: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Julian Lyndon-Smith wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? The lastest asterisk-addons I found at http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is cdr_addon_mysql.so. I've been using it with 1.2.2 and 1.2.3 without any problem other than the message during make install, which I just ignore. Is there a need for an update to asterisk-addons? ___ --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] Outgoing FXO and CDR
On Fr, 2006-01-27 at 14:06 +0100, Matt Riddell (IT) wrote: If you are the USA, you can try to use callprogress=yes in zapata.conf, but the warnings above the entry still stand. I will try callprogress but I thought it is just there for checking if the other side hung up? Also I'm not in USA does that mean callprogress will not work for me in Germany/Europe at all? Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --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] Nagios and Asterisk
Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. I was using bigbrother to do something similar I used wget to read from the status page, and detect colour changes to the status (like bbtray does), on a good/bad or bad/good change I dropped a call file in the callqueue directory that just played a canned wav file when the extension was picked up. since then changed to an email to sms gateway since it got too damn annoying, especially the fact the calls queued and repeated when unanswered Thanks, -- Darrell S. Long BestWeb Corporation ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.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
[Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Another piece to the puzzle, for what it's worth: The last moments before the crash, an incoming Zap call was answered by a SIP phone, parked, and then picked up by another SIP phone. During the picked up conversation, the audio was reported to me to be patchy, described as cell phone like. It is known that the calling party was not on a cell phone, but on a land line. ALSO, in the CDR CSV file, there is no mention of the call having been taken off park, as though the patchy call never happened. I confirmed this by looking at the specific Cisco phone, and seeing the last call that was made, and at what time. Does this speak to any suspect source of the issue? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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
RE: [Asterisk-Users] Re: Polycom 501 horrible echo
Can someone post the sample files somewhere for 1.6.2? I may have the same issue but the firmware dl from voipsupply I believe did not include the newer samples... Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Senykoff Sent: Saturday, January 28, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Polycom 501 horrible echo One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). That has got to be the problem! I'll let you know how the results go. Thanks ___ --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
[Asterisk-Users] Can't send DTMF transfer code from called SIP phone
I have several hardware and software phones connected to Asterisk 1.2.1 from Debian via SIP or IAX2 and I have defined call transfer codes in features.conf. Everything works with the only exception: When I call a _SIP_ _software_ phone (namely Ekiga or Kphone), I can't transfer the call from the _callee_ via the configured DTMF codes. It seems Asterisk completely ignores the sent DTMF codes (no transfer message is received and nothing is written on the log output). Transfer via the software phone transfer function works, as well as transfer via the DTMF codes when the SIP software phone acts as a caller. Any guess what can be wrong? Thanks for any advice. Milan Zamazal -- http://www.zamazal.org ___ --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] VOIP carriers and asterisk
I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? I think more information will be needed before someone can give you a useful reply. Things you might want to consider: 1) What country are you in? a local VoIP provider might give you better call rates in-country than a foreign provider. (this isn't always the case though - check carefully) 2) Where is the provider's network physically located? Do they route directly to/from the PSTN, or are there other upstream IP networks involved? 3) What latency do you get to their network from the box you're running asterisk on? 4) Do you require inbound numbers specific to your location? Again, this might limit your range of providers. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] VOIP carriers and asterisk
Con fecha 28/1/2006, burak balasaygun [EMAIL PROTECTED] escribió: Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? nufone.net voipjet.com voxee.com thanks Burak Balasaygu ___ --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] ISAC Codec Support
Erick Perez wrote: Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Which softphone supports it? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Best CoDec for high network latency
Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo. ___ --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] OT?: International number parsing
On Fri, January 27, 2006 23:47, Script Head said: What you're trying to accomplish can be easily done with an SQL query. You need to create a table of all the prefixes (international dial+country code+city/carrier) and join by that prefix. On 1/27/06, Damon Estep [EMAIL PROTECTED] wrote: Can anyone shed some light on rules that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another city code in the same country? I already have the table of country and city codes built. Are there holes in this theory; 1. Starting after the international dialing code, find the longest match for country code. 2. Starting after the country code from step 1, find the longest match for city code within that countries table of city codes. 3. The rest is the local number. Are there known exceptions? Am I reinventing the wheel rather than finding the right already existing resource? Obviously countrycodes are unique, and are created in a few 'classes' which also always provide unique numbers. Only one country has a single digit code: USA = 1 Most countries have a 2 digit code (31 = NL, 44 = UK, 49 = DE, etc.) There are *no* country codes with more than two digits that overlap the 2 digit codes. (So there's no 3 digit CC that starts with, for example, 31, 44, 49, etc.) So it is possible to 'categorize' them in to 1, 2, 3 digit CC's. Also the international dial codes have been chosen to not overlap anything else. So if you see (for instance) 011 you will always know it is an international call, and the next 1-3 digits will be a country code. -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?
I installed one and works fine but of course when I try to make the second call it says no lines are available That's weird. I was under the impression the non-Digium ones didn't care how many lines were in use, as there was no monitoring of such things in there. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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 150 SIP desktop phones with power over ethernet that will work with Plantronics HL-10 Handset Lifter for Remote Answering
I am looking for SIP with power over ethernet desktop phones that will work with asterisk and Plantronics HL-10 Handset Lifter for Remote Answering. Any suggestions? I am considering buying about 150 of these desktop phones for a new call center. ___ --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: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Warren, You may only use cdr_addon_mysql.so, but I believe that * normally automatically loads all modules it finds (see modules.conf for autoload=yes). The following modules were found in your modules directory, and 1.2.3 of * did not like them, because you got a warning after compile. In the case of app_rxfax.so and app_txfax.so these must of been compiled with a previous version of *, otherwise it would not have complained about them (I know this, because I had a similar issue). If you have kept the previous version of *, check your makefile for app_txfax and app_rxfax, make the same mods to your 1.2.3 makefile and recompile. * will then not complain about the *fax* modules. You may also need to recompile the asterisk-addons, simply because header files and or libraries may have changed in the core asterisk files. I guess what I am saying is that 1.2.3 of * may work with 1.2.1 of asterisk-addons (that is the latest version as you say), but asterisk-addons would need recompiling as well. If you make cleam;make and make install the asterisk-addons, do you get the same error when you make install asterisk ? Julian. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so Warren Burstein wrote: Julian Lyndon-Smith wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? The lastest asterisk-addons I found at http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is cdr_addon_mysql.so. I've been using it with 1.2.2 and 1.2.3 without any problem other than the message during make install, which I just ignore. Is there a need for an update to asterisk-addons? ___ --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: [Asterisk-Users] OT?: International number parsing
There is second single digit code - 7 (Russia). On Sat, 2006-01-28 at 18:41 +0100, Francesco Peeters (Asterisk) wrote: Only one country has a single digit code: USA = 1 ___ --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] Best CoDec for high network latency
Guillermo Salas M a écrit : Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? There are basically three parameters I can think of when speaking of voice over ip quality: 1 - Lag. In your case, a ping from your Asterisk box is 1 to 1.5 ms. Changing codecs is not going to help you here. 2 - Jitter. In your case, if the ping does vary between 1 and 1.5, that's 500ms ping jitter, which is high. You might want to have a large jitter buffer to compensate for it. But this increases lag even more... 3 - Packet drop. iLBC is meant to cope better with packet drop than other codecs, although in my experience any codec with too much packet drop will sound dreadful. If you have the bandwith and no packet loss, I would recommend that you bump up the jitter and stick with ulaw. While there might be a lot of lag - half duplex kind of conversations... - the audio should remain clear. If you are having packet loss on top of this, you might want to try iLBC... At any rate, nothing is going to replace trying out some settings for yourself... BTW: How come the latency is so high? The worst I've seen so far was a link varying between 600 and 1200ms and the quality varied from good enough to pretty horrible... Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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 150 SIP desktop phones with power over ethernet that will work with Plantronics HL-10 Handset Lifter for Remote Answering
On 10:21, Sat 28 Jan 06, Nilesh Londhe wrote: I am looking for SIP with power over ethernet desktop phones that will work with asterisk and Plantronics HL-10 Handset Lifter for Remote Answering. Any suggestions? I am considering buying about 150 of these desktop phones for a new call center. Don't know any good phones. But I do know you don't want to buy the snom 190 (now 200) for this. The remote lifter doesn't fit. So unless you ok with taping it together, dont use the snom190 together with the HL-10 -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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
[Asterisk-Users] How to Unregister?
Hi all, When i am using database show command, i can see more than 100 users are registered but actually they are not 100 some IP Phones are continue registered even i closed and switch off the IP Phone. Actually i am doing Windows based GUI, so i want to display all real registered users. I am using mySQL relatime for authuntication. I will be appriciate if any one can tell me how i can unregister so i will make some code to do unregisteration which ip phones are not registered. I will be appriciate for your replys. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ 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
[Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
It looks like my modules are all up to date - they are all dated the 25th Jan - aka Black Wednesday. What really bugs me about this is the lack of useful info from any logs. The last call to take place, the call that gets distorted, has no entry. This has to indicate something, no? Hmm - I'd do as others have suggested and move the /usr/lib/asterisk/modules directory to another, and do a make clean;make;make install If you have app_rxfax.so installed then you must have customised your original makefile, and not the 1.2.3 makefile, which would suggest that these modules are from a previous asterisk version. Let us know how you get on. Julian. Dan Littlejohn wrote: On 1/27/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? If * is loading older modules (which it probably is because of your config files) then it may cause grief ;) My .2p worth. Probably not helpful, but maybe, just maybe Julian Dan Littlejohn wrote: On 1/27/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Brent - Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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
Re: [Asterisk-Users] regarding connecting to AMP
http://mundy.org/blog/index.php?p=93 http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki (Chapter 4 and 7) The above links have some excellent documentation. www.voip-info.org specifically has some really good setup examples. Recommend you go through those... -R Sohail Arham wrote: hi alli have intalled [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] successfully and now the problem is that how can i connect to AMP so that i would be able to configure it.actually i have following setup... one [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] machine and two other machines i want that these two clients machine can be able to call each other through using [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] box.i connect this [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] box to the hub...(simple hub) ...now tell me what ip scheme i would use to configure it ...and how it would be possible to complete my task...one more thing i have also xlite sip phone ...i will call these two machine through these sip soft phonesnow plz temme complete idea becaz i have no good experience about it.i shall be thankful to you BYE -- Muhammad Sohail Arham U.E.T. Lahore Phone No. 0321-4422406 ___ --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
[Asterisk-Users] Re: (Un)PauseQeueMamber usage
Thanks for the reply BJ. Your example makes sense for out-bound traffic, but
what about calls transferred from a queue to an agent?
I plan on setting up agent extensions (if possible via macro) something like
this for example:
exten = 1234,1,PauseQueueMember (|Agent/101)
exten = 1234,2,Dial(Agent/101,tg)
exten = 1234,3,UnPauseQueueMemeber(|Agent/101)
exten = 1234,4,Hangup()
Agents will login using AgentCallBackLogin. In the example above, Agent 101
will login from extension 1234. This would work well if Agent 101 was always
sitting at the phone with extension 1234. This will more than likely not be
the case.
Is this what I need:
exten = 1234,1,PauseQueueMember(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,2,Dial(Agent/${AGENTBYCALLERID_${CALLERIDNUM}},tg)
exten =
1234,3,UnPauseQueueMemeber(|Agent/${AGENTBYCALLERID_${CALLERIDNUM}})
exten = 1234,4,Hangup()
Not sure if this is the proper use of this variable or not.
Regards to all,
Joe
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Re: [Asterisk-Users] Simple question about ringing multiple phones (extensions)?
On Jan 28, 2006, at 12:54 AM, Ronald Wiplinger wrote:
Martin Joseph wrote:
snipI tried something like:
exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL=${PHONE_601}${PHONE_602}${PHONE_603}
PHONE_601=SIP/601; office 601 Ronald
PHONE_602=SIP/602; office 602 Ronald
PHONE_603=ZAP/1r1; living room 603 cordless
For you this should work too:
exten = 2020,2,Dial(SIP/2005IAX/2010,25,tr)
Thanks very much for the help guys!
Marty
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RE: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2?
Ok I've just shot an email off to my service provider to confirm I can make more than 1 g729 call at a time. Can anyone in here confirm that the non-digium lines don't care how many calls you are making at a time. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Saturday, 28 January 2006 12:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2? I installed one and works fine but of course when I try to make the second call it says no lines are available That's weird. I was under the impression the non-Digium ones didn't care how many lines were in use, as there was no monitoring of such things in there. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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: [Asterisk-Users] Best CoDec for high network latency
Con fecha 28/1/2006, Jean-Michel Hiver [EMAIL PROTECTED] escribió: Guillermo Salas M a écrit : Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? There are basically three parameters I can think of when speaking of voice over ip quality: 1 - Lag. In your case, a ping from your Asterisk box is 1 to 1.5 ms. Changing codecs is not going to help you here. The lag if 1000 ~ 1500 ms 2 - Jitter. In your case, if the ping does vary between 1 and 1.5, that's 500ms ping jitter, which is high. You might want to have a large jitter buffer to compensate for it. But this increases lag even more... 3 - Packet drop. iLBC is meant to cope better with packet drop than other codecs, although in my experience any codec with too much packet drop will sound dreadful. If you have the bandwith and no packet loss, I would recommend that you bump up the jitter and stick with ulaw. While there might be a lot of lag - half duplex kind of conversations... - the audio should remain clear. I don't have packet loss, but my BW is limited. If you are having packet loss on top of this, you might want to try iLBC... At any rate, nothing is going to replace trying out some settings for yourself... BTW: How come the latency is so high? The worst I've seen so far was a link varying between 600 and 1200ms and the quality varied from good enough to pretty horrible... Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ --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] Voip Provider
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote: x-tad-smallerHi Everyone,/x-tad-smallerx-tad-smallerI know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100’s to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service./x-tad-smallerx-tad-smallerI use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. I’ve been looking for service for 2 months and I haven’t had any luck./x-tad-smallerx-tad-smallerP.S. I do not need any special services, just proper DTMF tone handling./x-tad-smallerThis might be a codec negotiation issue with the termination service. I am using Teliax with my asterisk server to terminate my SIP and IAX calls from several ATAs and softphones. All of that works fine with DTMF. I am using the G729 codec exclusively for my Teliax calls. You also need to be sure that the extensions for each ATA/phone have the DTMF configured righteously. HTH, Marty ___ --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] Installing the none commercial intelg729codecsinto [EMAIL PROTECTED] 2.2?
I'll confirm this. I've been using the non-digium g729 codecs for some time now. During testing, we had about 15 calls going at once (using Teliax and voipjet). Codecs didn't seem to care how many calls were going. I didn't do anything special; just put the codec file in the folder with all the other codecs and restarted the server. Sorry I cant be of more help! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, January 28, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Installing the none commercial intelg729codecsinto [EMAIL PROTECTED] 2.2? Ok I've just shot an email off to my service provider to confirm I can make more than 1 g729 call at a time. Can anyone in here confirm that the non-digium lines don't care how many calls you are making at a time. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Saturday, 28 January 2006 12:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2? I installed one and works fine but of course when I try to make the second call it says no lines are available That's weird. I was under the impression the non-Digium ones didn't care how many lines were in use, as there was no monitoring of such things in there. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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: [Asterisk-Users] Re: (Un)PauseQeueMamber usage
Joe wrote: Thanks for the reply BJ. Your example makes sense for out-bound traffic, but what about calls transferred from a queue to an agent? When an agent receives a call, they will be marked busy anyways as long as you are using agent members for the queue. (member = Agent/1000) Kevin ___ --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] Best CoDec for high network latency
You need to use the 'over' codec. It has been used for years with half-duplex conversations. OVER . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Saturday, January 28, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best CoDec for high network latency Guillermo Salas M a écrit : Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? There are basically three parameters I can think of when speaking of voice over ip quality: 1 - Lag. In your case, a ping from your Asterisk box is 1 to 1.5 ms. Changing codecs is not going to help you here. 2 - Jitter. In your case, if the ping does vary between 1 and 1.5, that's 500ms ping jitter, which is high. You might want to have a large jitter buffer to compensate for it. But this increases lag even more... 3 - Packet drop. iLBC is meant to cope better with packet drop than other codecs, although in my experience any codec with too much packet drop will sound dreadful. If you have the bandwith and no packet loss, I would recommend that you bump up the jitter and stick with ulaw. While there might be a lot of lag - half duplex kind of conversations... - the audio should remain clear. If you are having packet loss on top of this, you might want to try iLBC... At any rate, nothing is going to replace trying out some settings for yourself... BTW: How come the latency is so high? The worst I've seen so far was a link varying between 600 and 1200ms and the quality varied from good enough to pretty horrible... Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
Hi Ross, thanks for this. It appears there was some problem when asterisk went from 1.07 to 1.2 The non-commercial codec providers are aware of this but dont know when/if they will be able to fix this (this obviously also affects anyone who is running [EMAIL PROTECTED] 2.0 and up) Guess if it's a big enough problem buy a commercial codec etc. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross C Sent: Saturday, 28 January 2006 3:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2? I'll confirm this. I've been using the non-digium g729 codecs for some time now. During testing, we had about 15 calls going at once (using Teliax and voipjet). Codecs didn't seem to care how many calls were going. I didn't do anything special; just put the codec file in the folder with all the other codecs and restarted the server. Sorry I cant be of more help! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, January 28, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Installing the none commercial intelg729codecsinto [EMAIL PROTECTED] 2.2? Ok I've just shot an email off to my service provider to confirm I can make more than 1 g729 call at a time. Can anyone in here confirm that the non-digium lines don't care how many calls you are making at a time. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Saturday, 28 January 2006 12:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2? I installed one and works fine but of course when I try to make the second call it says no lines are available That's weird. I was under the impression the non-Digium ones didn't care how many lines were in use, as there was no monitoring of such things in there. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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 ___ --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: no progress indications on isdn phone connected to capi card (was: using a Gigaset SX440isdn on a Diva 4BRI?)
On Fri, 13 Jan 2006, Louis-David Mitterrand wrote: On Fri, Jan 13, 2006 at 02:03:20PM +0100, Armin Schindler wrote: On Wed, 11 Jan 2006, Louis-David Mitterrand wrote: On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote: On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote: On Tue, 10 Jan 2006, Louis-David Mitterrand wrote: [C:4] 22:0188:202 - D-X(003) 02 01 7F [C:4] 22:0189:202 - D-X(003) 02 01 7F [C:4] 22:0190:202 - D-X(003) 02 01 7F [C:4] 22:0191:201 - MDL-ERROR(G) [C:4] 22:0191:202 - SIG-EVENT 0A The diva card is sending (D-X), but does not receive anything (D-R). It looks like either the cross connection still isn't working or the protocol is wrong. OK, making some progress here: I removed -u (ptp mode) from the divactrl init string and now I can call in and out with my Gigaset handset! Calling and receiving calls works but I get no call progress indications at all until the call is connected. Even when using immediate=yes and landing directly in exten = s,1,Dial(CAPI/g2//bo) I get no dial tone. Is there some setting I forgot about? What version of card/driver/protocol-code do you use? pyrrhus:~# divactrl ctrl -c 1 -CardInfo 0xfe7a9f00 0xce00 0xfd00 0xfe7b 0x 0x 0x 0x 0x16 pyrrhus:~# divactrl ctrl --version divaload, BUILD (local[102-52]-Sep 27 2005-17:18:38) pyrrhus:~# divactrl ctrl -c 1 -CardName Diva Server 4BRI-8M 2.0 PCI pyrrhus:~# l /usr/share/eicon/te_etsi.* -rw-r--r-- 1 201 200 691696 2003-11-12 15:49 /usr/share/eicon/te_etsi.qm0 -rw-r--r-- 1 201 200 691696 2003-11-12 15:49 /usr/share/eicon/te_etsi.qm1 -rw-r--r-- 1 201 200 691696 2003-11-12 15:49 /usr/share/eicon/te_etsi.qm2 -rw-r--r-- 1 201 200 691696 2003-11-12 15:49 /usr/share/eicon/te_etsi.qm3 -rw-r--r-- 1 201 200 583968 2003-11-12 15:49 /usr/share/eicon/te_etsi.sm -rw-r--r-- 1 201 200 398020 2003-11-12 15:49 /usr/share/eicon/te_etsi.sm.4 chan_capi is today's CVS version. Please create a verbose log level 5 with capi debug. It maybe because of an older version of protocol code. Please find the log attached. I didn't find any attached file. Anyway you would need to use new driver/firmware from eicon source RPM. (Melware will soon provide new driver V3 with full support) Armin ___ --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] Help with Music on Hold during transfer
Hi, Does anyone have an extensions.confscript example showing the following:- 1) Incoming call is answered. 2) Incoming caller is played a looping welcome message until step 4. 3) A call is placed to an extension/psdn. 4) When call is answered, the music ends and the incoming caller is transfered to the outgoing call from step 3. Thank you Dan Journo www.textover.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
Re: [Asterisk-Users] DTMF's indescipherable, but voice clean!
Maybe the company providing the DID so other people can say if they arehaving problems, and the SIP debug on a call that has problems. I think that's it actually. I'm using sipphone.com (Gizmoproject) and I called myself from the Gizmo client and experienced the problem. Perhaps I should move this thread to their forum. Gabe ___ --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] Best CoDec for high network latency
Alexander Lopez a écrit : You need to use the 'over' codec. It has been used for years with half-duplex conversations. OVER Yeah, right. 73, 51 :) Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] AutoDialing with VOP USING SIPURA 2100'S
Hello all, I am trying to find out if anyone has a provider that is good with dtmf playback using a Sipura 2100? Ive just dialed with voxee and the call goes through but when I press 1 my dialer does not hear it. My dialer is making the call using a Dialogic d/4PCI connected to the Sipura 2100 through voxee and I am calling my landline. When I pick up the landline and say hello it properly carries the call except when I have to press 1 to continue with the call voxee is not sending it back the right way. Oddly enough I just set up a teliax account and did the same thing and it worked great. This tells me that it is server side if it works with one provider and not the other. The problem with teliax is that im not paying 2 cents per call essentially to deliver a 45 second message. For those that dont know, teliax offers termination at 2 cents per minute 60/1 Bottom line teliax service with voxee pricing would be great if anyone has tested any providers that properly relay touchtones. Mark Adams ___ --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] RoadRunner
Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit ___ --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] voicetronix FXOs with * ?
Anyone used voicetronix FXOs with * ? I'm interested to know how they compare with eg TDM400P. Specifically I'm interested in how good the echo canceller is. -Dan ___ --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] RoadRunner
I use SIP over VPN with RR from TWC no problem, connect via WiFi. According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps down and 353Kbps up at this time (6:15pm Saturday). My laptop currently has an X-Lite (free version) softphone with GN Netcom USB professional contact center headsets (GN8110 USB XP adapter). We have found that the headset makes a major difference in the quality of the results. Bill Hunt Stroudwater Contact Point -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen Sent: Saturday, January 28, 2006 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RoadRunner Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit ___ --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
[Asterisk-Users] english snom support forums ?
Is there a forum for snom support in english? There are some very active snom forums but they appear to be entirely german language only. -Dan ___ --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] RoadRunner
Yep I use iax and sip with time warner cable new york. Works fine. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen Sent: Saturday, 28 January 2006 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RoadRunner Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit ___ --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
[Asterisk-Users] Multiple Subscriptions to SIP accounts at Same Domain
Sorry not to have observed etiquet and lurked here for a bit before wading in with a question but I have an issue that may well be because I dont know enough about what asterisk is actually doing under the hood to understand why I cant do what I want with asterisk. Im hoping that someone can point me in the right direction :-) This is what I have: Mandrake 2006 running Asterisk 1.2.3 - no additional hardware - everything is going to be running via SIP. To enable inbound and outbound connectivity I have been experimenting with using various accounts provided by Gosspitel, Sipgate, aql and others and have found the most sucessful have been those provided by Gossiptel. Herein lies the problem. I need to register about six incoming lines all provided by Gossiptel - half of them to be active within one context and half within another. I have sucessfully registered all the lines within sip.conf as follows: register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] etc and then I created a peer and a user for the sip.gossiptel.com domain, but I now find that any calls that come in to any of these registered accounts all ring the 's' extension within the default context. Thats fine as far as it goes but I need to be able to handle each SIP account in its own context. As a half way house, in the course of testing this I did play with creating extensions for each sip account and directing them thus: register = username1:password1:[EMAIL PROTECTED]/ext1 register = username2:password2:[EMAIL PROTECTED]/ext2 and this works fine as well - inbound calls end up activating the assigned extensions within extensions.conf but the problem remains that these extensions themselves have to be within a single context (in my case the default context). So my question in short is - does anyone know how I can regester multiple SIP accounts so that, at the time of registration they each become active in different contexts - I have scoured the manual and see no way of allocating contexts to each individual registration if they are all at the same domain. OK, Im labouring the point now because its late :-) I would have thought that one way of doing this would be to have some way of forcing a peer and user definition to register with the server and for that registration to be within the context of the peer or user - so that inbound calls to that line would activate the extension within the assigned context - or perhaps to have a context switch that could be procesed by register = wherever it was in sip.conf I would have thought the former solution would be better as it seems more logical and understandable to put this within a peer definition - does anyone know if simply putting the register = command in a peer definition has the desired effect ? Certainly it didnt seem to when I tried it. Hope that this is clear enough and that someone has the answer to this one! Its quite annoying as to my mind it should clearly be possible, especially for those of us who need multiple lines and want to stick to one voip provider externally. Kind Regards and thanks in advance for any help with this one, Geoff. ___ --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] RoadRunner
Dean Collins wrote: Yep I use iax and sip with time warner cable new york. Works fine. Dean IAX2 and SIP used here. All systems green. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] RoadRunner
Hello, We have used RoadRunner for the last 3 years to conect several of our offices(IAX2-VOIP and data) and it was great up until about 6 months ago. At that time we started having random outages and horrible latency bumps at all of our offices. RR acknowledges the outages, but they haven't stopped. We still have several a week(from 5 minutes to 12 hours) and we have gotten a lot of credit to our account because of the outages, but it just isn't worth it anymore. Something seriously wrong must've happend to the RR architecture here in the Tampa Bay area about 6 months ago and they cannot/will not fix the problem. We've had over a dozen different network engineers from RR out to our various locations and they say that's just the way it is. RR does not offer any kind of SLA on any of their Cable internet connections so there is really no recourse other than to complain or drop service(which is what we are going to do in about 3 months) We will be evaluating point-to-point data T1s and a newer technology RSair wireless internet(where they put a tower on your building) starting next month and neither of those has the shared-network and infrastructure issues of cable or DSL so I am hopeful it will work for us. Your experience depends entirely on the quality of the Cable infrastructure in your area and whether your neighbors like to hog the bandwidth in your neighborhood. Good luck. MATT--- On 1/28/06, Rene Kluwen [EMAIL PROTECTED] wrote: Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit ___ --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: [Asterisk-Users] Multiple Subscriptions to SIP accounts at SameDomain
Original Message From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 1:29 AM Subject: [Asterisk-Users] Multiple Subscriptions to SIP accounts at SameDomain Sorry not to have observed etiquet and lurked here for a bit before wading in with a question but I have an issue that may well be because I dont know enough about what asterisk is actually doing under the hood to understand why I cant do what I want with asterisk. Im hoping that someone can point me in the right direction :-) This is what I have: Mandrake 2006 running Asterisk 1.2.3 - no additional hardware - everything is going to be running via SIP. To enable inbound and outbound connectivity I have been experimenting with using various accounts provided by Gosspitel, Sipgate, aql and others and have found the most sucessful have been those provided by Gossiptel. Herein lies the problem. I need to register about six incoming lines all provided by Gossiptel - half of them to be active within one context and half within another. I have sucessfully registered all the lines within sip.conf as follows: register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] etc and then I created a peer and a user for the sip.gossiptel.com domain, but I now find that any calls that come in to any of these registered accounts all ring the 's' extension within the default context. Thats fine as far as it goes but I need to be able to handle each SIP account in its own context. As a half way house, in the course of testing this I did play with creating extensions for each sip account and directing them thus: register = username1:password1:[EMAIL PROTECTED]/ext1 register = username2:password2:[EMAIL PROTECTED]/ext2 and this works fine as well - inbound calls end up activating the assigned extensions within extensions.conf but the problem remains that these extensions themselves have to be within a single context (in my case the default context). From sip.conf: ;register = [EMAIL PROTECTED]/1234 ; ;Register 2345 at sip provider. Calls from this provider connect to local ;extension 1234 in extensions.conf default context, unless you define ;[mysipprovider.com] in a section below, and configure a context Wild guess: A kludge is if you run your own dns: *.gossiptel.mydom.dom.INCNAMEsip.gossiptel.com. Then register each user to his own domain: register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] Then define [username1.gossiptel.mydom.dom] context=user1context [username2.gossiptel.mydom.dom] context=user2context Otherwise, you should just create a patch to allow the syntax register = user[:secret[:[EMAIL PROTECTED]:port][/context[/extension]] Shouldn't be so hard to do :-) Leif ___ --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] Multiple Subscriptions to SIP accounts at Same
register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] etc and then I created a peer and a user for the sip.gossiptel.com domain, but I now find that any calls that come in to any of these registered accounts all ring the 's' extension within the default context. change the context within sip.conf to from-sip-provider or something like that. Thats fine as far as it goes but I need to be able to handle each SIP account in its own context. use extensions.conf for this purpose (we did). in sip.conf you have: register = username1:password1:[EMAIL PROTECTED]/ext1 register = username2:password2:[EMAIL PROTECTED]/ext2 then in extensions.conf you have [from-sip-provider] exten = ext1,1,Goto(context-for-ext1,s,1) exten = ext2,1,Goto(context-for-ext2,s,1) As a half way house, in the course of testing this I did play with creating extensions for each sip account and directing them thus: so you were halfway there and this works fine as well - inbound calls end up activating the assigned extensions within extensions.conf but the problem remains that these extensions themselves have to be within a single context (in my case the default context). that's the dialplan's problem - to sort it all out. :-) note that we're doing this with dozens of numbers with no problem. as a possibly helpful hint, it is nice to include the phone number as part of the extension, such as ext4148441414 or did4148441414 rather than ext1. there may be some downsides to using just the number by itself; it's been a while and i don't recall for sure. it seems like there should be a way to make this work within sip.conf itself, but the interactions between the registrations and definitions has always seemed to be loose at best and i've never been able to get them to work the way i would expect, so beware that other more correct solutions may exist. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] Multiple Subscriptions to SIP accounts at Same
On 18:57, Sat 28 Jan 06, Joe Greco wrote:
register = username1:password1:[EMAIL PROTECTED]
register = username2:password2:[EMAIL PROTECTED]
etc
and then I created a peer and a user for the sip.gossiptel.com domain,
but I now find that any calls that come in to any of these registered
accounts all ring the 's' extension within the default context.
change the context within sip.conf to from-sip-provider or something
like that.
Thats
fine as far as it goes but I need to be able to handle each SIP account
in its own context.
use extensions.conf for this purpose (we did).
in sip.conf you have:
register = username1:password1:[EMAIL PROTECTED]/ext1
register = username2:password2:[EMAIL PROTECTED]/ext2
then in extensions.conf you have
[from-sip-provider]
exten = ext1,1,Goto(context-for-ext1,s,1)
exten = ext2,1,Goto(context-for-ext2,s,1)
As a half way house, in the course of testing this
I did play with creating extensions for each sip account and directing
them thus:
so you were halfway there
and this works fine as well - inbound calls end up activating the
assigned extensions within extensions.conf but the problem remains that
these extensions themselves have to be within a single context (in my
case the default context).
that's the dialplan's problem - to sort it all out. :-)
note that we're doing this with dozens of numbers with no problem. as a
possibly helpful hint, it is nice to include the phone number as part of
the extension, such as ext4148441414 or did4148441414 rather than
ext1. there may be some downsides to using just the number by itself;
it's been a while and i don't recall for sure.
We don't use SIP but IAX instead.
Protocol doesn't matter in this case.
It all boils down to extensions.conf magic.
Hell, we even get all our numbers using only 1 account.
This is a snippet from our extensions.conf:
CUST001DID = a
CUST002DID = b
[incoming-from-provider]
exten = XXXa,1,Goto(CUST001,${EXTEN},1)
exten = XXXb,1,Goto(CUST002,${EXTEN},1)
etc etc etc
Works like a charm for multiple customers on 1 asterisk
cluster.
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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[Asterisk-Users] Urgent: Unable To Execute after updating from SVN
Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] = (Scan Zap channels application) == Registered application 'ZapScan'[app_saycountpl.so] = (Say polish counting words) == Registered application 'SayCountPL'[func_cut.so] = (Cut out information from a string) == Registered custom function CUT == Registered custom function SORT[app_echo.so] = (Simple Echo Application) == Registered application 'Echo'[app_alarmreceiver.so] = (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver'[app_settransfercapability.so] = (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [app_url.so] = (Send URL Applications) == Registered application 'SendURL'[app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_md5.so: undefined symbol: option_priority_jumping Jan 29 02:49:10 WARNING[32424]: loader.c:555 load_modules: Loading module app_md5.so failed! Any ideas? Thanks Dan Journo www.TextOver.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
[Asterisk-Users] Extension vs. Mailbox numbers and an example of an AGI script...
I have users with multiple extensions (ie. office, home, etc.) and I
wanted my users to have only one mail box for all of these extensions.
Perhaps I didn't see it in the Asterisk docs and I have bastardized the
sip.conf but I did the following. Constructive flames appreciated.
Tim
In sip.conf the mailbox line shows the common mailbox number (ie 1015)
with the optional context:
[...]
[1015]
type=friend
username=1015
callerid = Tim Pozar 1015
secret=HACKME
host=dynamic
nat=yes
context=default
canreinvite=no
mailbox=1015
[1078]
type=friend
username=1078
secret=HACKME
host=dynamic
nat=yes
context=default
canreinvite=no
[EMAIL PROTECTED]
callerid=Tim Pozar HOME 1078
[...]
In extesions.conf I run an AGI that goes out and gets the mailbox
numberr and passes it to the Voicemail and VoicemailMain commands...
[...]
exten=_1XXX,1,dial(SIP/${EXTEN},20,tr)
exten=_1XXX,2,Wait,1
exten=_1XXX,3,agi(parse_asterisk_sip_conf.agi|${EXTEN}|mailbox)
exten=_1XXX,4,Voicemail(u${MAILBOX})
exten=_1XXX,5,Hangup
exten = 8500,1,Answer
exten = 8500,2,Wait,1
exten = 8500,3,agi(parse_asterisk_sip_conf.agi|${CALLERIDNUM}|mailbox)
exten = 8500,4,VoicemailMain(${MAILBOX})
exten = 8500,5,Hangup
[...]
parse_asterisk_sip_conf.agi looks like this...
---
#!/usr/bin/perl
use strict;
#
# Parses the sip config to pass back the mailbox number.
# There should be more error checking in this script.
#
# Tim Pozar - Sat Jan 28 18:22:40 PST 2006
#
$|=1;
my $asterisk_conf_dir = /etc/asterisk;
my $sip_conf = $asterisk_conf_dir/sip.conf;
my $value = ;
my $sip_extennum = $ARGV[0];
my $sip_variable = $ARGV[1];
my $variablename;
my %AGI;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
open(CONF, $sip_conf) || die can't open $sip_conf;
while (CONF) {
chop;
if ((!/^;/) (!/^#/)) { # Skip comments at the start of lines
if (/^\[$sip_extennum\]/) {
$_ = CONF;
chop;
while ((!/^\[/)){
if (/^$sip_variable/) {
($variablename, $value) = split('=');
last;
}
$_ = CONF;
chop;
}
}
}
}
# Make sure you pass back something and not just a blank. Asterisk
# doesn't like that.
$variablename = uc($variablename);
print SET VARIABLE $variablename $value\n\n;
exit 0
---
--
1978 45th Ave / San Francisco CA 94116 / USA // POTS: +1 415 665 3790
GPG Fingerprint: 4821 CFDA 06E7 49F3 BF05 3F02 11E3 390F 8338 5B04
Life is playful - Ben Olizar
begin:vcard
fn:Tim Pozar
n:Pozar;Tim
org:UnitedLayer LLC
adr:Suite 110;;200 Paul Avenue;San Francisco;CA;94124-3100;US
email;internet:[EMAIL PROTECTED]
title:COO
tel;work:415-349-2112
tel;home:415-665-3790
tel;cell:415-637-8512
note:Be who you are and say what you feel because the people who mind don't matter and the people who matter don't mind. - Dr. Seuss
url:http://www.unitedlayer.com
version:2.1
end:vcard
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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
[Asterisk-Users] Problem with MusicOnHold
Hi, Does anyone know what this means? Jan 29 03:17:42 WARNING[6276]: format_mp3.c:158 mp3_squeue: Short read (-1) (Bad file descriptor)!Jan 29 03:17:42 WARNING[6276]: format_mp3.c:158 mp3_squeue: Short read (-1) (Bad file descriptor)! Thanks Dan Journo www.TextOver.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
Re: [Asterisk-Users] RoadRunner
Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? I have an employee using a Cisco 7960 over RoadRunner, 15 hops away, working just fine with g711. Some cable companies are known to use rate-limiting devices to reduce inbound/outbound Internet traffic. You might ask their tech support folks if they are using such as box. ___ --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] RoadRunner
Vote with your feet and go elsewhere, sla or not it wont take them long to revisit the situation. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Saturday, 28 January 2006 7:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RoadRunner Hello, We have used RoadRunner for the last 3 years to conect several of our offices(IAX2-VOIP and data) and it was great up until about 6 months ago. At that time we started having random outages and horrible latency bumps at all of our offices. RR acknowledges the outages, but they haven't stopped. We still have several a week(from 5 minutes to 12 hours) and we have gotten a lot of credit to our account because of the outages, but it just isn't worth it anymore. Something seriously wrong must've happend to the RR architecture here in the Tampa Bay area about 6 months ago and they cannot/will not fix the problem. We've had over a dozen different network engineers from RR out to our various locations and they say that's just the way it is. RR does not offer any kind of SLA on any of their Cable internet connections so there is really no recourse other than to complain or drop service(which is what we are going to do in about 3 months) We will be evaluating point-to-point data T1s and a newer technology RSair wireless internet(where they put a tower on your building) starting next month and neither of those has the shared-network and infrastructure issues of cable or DSL so I am hopeful it will work for us. Your experience depends entirely on the quality of the Cable infrastructure in your area and whether your neighbors like to hog the bandwidth in your neighborhood. Good luck. MATT--- On 1/28/06, Rene Kluwen [EMAIL PROTECTED] wrote: Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit ___ --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: [Asterisk-Users] Urgent: Unable To Execute after updating from SVN
When you did the make install of asterisk, it gave you a whole pile of modules it didnt know about, some of which no longer work with trunk. The easiest way to fix this: # mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.old then do a make install again --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Journo Sent: Sunday, 29 January 2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Urgent: Unable To Execute after updating from SVN Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] = (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] = (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] = (Cut out information from a string) == Registered custom function CUT == Registered custom function SORT [app_echo.so] = (Simple Echo Application) == Registered application 'Echo' [app_alarmreceiver.so] = (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' [app_settransfercapability.so] = (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [app_url.so] = (Send URL Applications) == Registered application 'SendURL' [app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_md5.so: undefined symbol: option_priority_jumping Jan 29 02:49:10 WARNING[32424]: loader.c:555 load_modules: Loading module app_md5.so failed! Any ideas? Thanks Dan Journo www.TextOver.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
[Asterisk-Users] Adjusting gain, Milliwatt and ztmonitor
I have been trying to adjust the gain as per this document without any success: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html I have a PSTN and VoIP (SIP) connection via *. I disabled all echo cancel/training in zapata.conf and set tx/rxgain to 0. I then changed my extensions.conf so that when I call the VoIP line from the PSTN line, it plays the Milliwatt application tone. However, when I call, I don't hear the tone and ztmonitor doesn't change at all. What could be the problem? I am running the latest svn source, Aastra 9133i sip phone and Digium TDM11B. BTW - My PSTN is thru Verizon (MA) and I don't have their test tone #. If you know it, please e-mail it to me. ___ --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] 5,000 concurrent calls system rollout question
On Jan 28, 2006, at 7:15 PM, Vic wrote: Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? You can also use it as and end use agent (ie an ATA or a phone). I am using an AG-168V which is a cheapo ATA that supports IAX directly. This is nice because it simplifies ports and firewall issues. In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, can we make it a bt more feasible number? Transcoding is a big consumer of CPU for sure. This has nothing to do with SIP however and is related to the CODEC you are using at the end of the line and in between. If all you calls are coming in and being delivered in the same format (ie g729), then you don't need to transcode anything, and the CPU load is much lighter. In fact you can setup asterisk to make a native bridge of these calls. Perhaps you could try building a testbed? That's what I would do. Good Luck, Marty Zoa [EMAIL PROTECTED]> wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: > Hi, > > we are currently considering different options for rolling out a large > scale IP PBX to handle around 3,000 + concurrent calls. > > Can this be done with Asterisk? Has it been done before? > > I really would like an input on this. > > Thanks! > > > >___ >--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 ___ --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] RoadRunner
On Jan 28, 2006, at 2:13 PM, Rene Kluwen wrote: Is somebody here using a RoadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Run some trace routes from your IP to your VOIP host, and then call Road runner and complain. This kind of latency isn't very unusual for them or any cable modem provider in my experience, but if you keep complaining and showing them the problem, they MIGHT fix it. Marty ___ --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] 5,000 concurrent calls system rollout question
The question is somewhat ludicrous, and Im slightly surprised that no-one has sat down and done the maths about bandwidth utilization. So I did. To handle 5000 calls coming in over a PRI, youd need 210 or so T1s or 170 E1s. All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) There is no way possible that youre going to pump that amount of data through a PC. Dont care about codecs and dialplans, PCs just dont have that sort of internal bandwidth from peripherals. If you do, honestly, need to handle 5k calls, youd probably have to have a bank of Cisco 5850s doing the termination With a max of 5 DS3 (1 DS3 = 28 T1s) into each one, youll need 8, or probably 9 as youd want to have one as a hot spare. Each of those DS3s would go into some beefy switching fabric (note, that each one is producting 225mbit) and youd have some sort of asterisk box with huge internal bandwidth handling each one. Cross connect all 9 asterisk boxes via 10Gbit networks (note, youll need PCI-16x 10g cards) and have a pair of voicemail servers. Id suggest a pair of big Sun boxes. Then, of course, you have the issue of getting the calls _out_ of the asterisk machines. Youve just doubled your bandwidth requirements, so youll need to double up on the asterisk machines, and split the network up further. Id take a guess that you could do it under USD$1million (just for hardware) but I wouldnt be surprised if it was USD$10million. Im happy to sell you any of this 8-) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Sunday, 29 January 2006 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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
