RE: [Asterisk-Users] QOS Device?
Seems interesting enough. I have two questions. a. what are you running on Fedora Core to shape the traffic? b. let's say that you have VPN site to site tunnels from the FW behind the QoS machines towards a branch office and that some of the traffic in the Tunnel has higher priority then other traffic. The QoS device sees it all as encrypted traffic and can't help there. What would you suggest? would placing the QoS machines elsewhere help? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 9:02 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] QOS Device? I will be putting documentation together shortly on how to build a high-availability QoS setup using 2 spare PCs and 4 NICs. I've been very successful with this approach for a T-1 that shares both Citrix and Video Conferencing + normal web traffic and such. The real key is a combination of packet prioritization with traffic shaping. The QoS boxes I build use Fedora Core 1 and are configured as bridges. This way, you just drop them into the right spot on the network and don't have to change routes or anything. Also, I put ntop on them, so they can monitor traffic statistics to/from the WAN. They use Spanning Tree Protocol (part of the bridge-utils package) to make the solution high availability. All traffic routes through the primary QoS box, but if it fails traffic goes through the second box. I took this approach because I was using old HP Vectras (Pentium 200 Pros) that have old drives in them, which _will_ fail at some point. The Vectras were just sitting on the shelf, and I've got more customized shaping going on than any cookie cutter solution will give you. Here's a simple diagram: - | T-1 | - | --- | switch | --- | | | | -- -- |QoS1| |QoS2| -- -- | | | | --- | switch | --- | | firewall | | | --- --- | LAN | | DMZ | --- --- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Capi, Controller
On Thursday 16 December 2004 06:59, SIN - Robert Siedl wrote:
Hi List,
I have asterisk 1.0 on a SuSe Linux 9.0 with one AVM C4 ISDN card an one
AVM Fritz card running for outgoing an incomming calls.
From austria telecommunications company I have two isdn nt, the connectet
on
avm c4 card and I have one gsm-gateway for mobile handy.
How can I asterisk instill, wenn a outgoing call beginn with 0664, take the
controller 3 (=avm fritz card) else take controller 1 or 2 (avm c4 card on
telekom nt)
Have somebody a idea?
Hi,
you control which controller a call goes out by specifying the relevant msn.
configs something like (of top of my head):
capi.conf
[interfaces]
msn =12345
controller=1
msn = 54321
controller=3
extensions.conf
exten = _0664.,1,dial(capi/54321:${EXTEN})
exten = _.,1,dial(capi/12345:${EXTEN})
first line would send call out of controller 3, 2nd line sends it out
controller1 .
HTH
Jon
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Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
Being optimistic, I think it's a great idea.. putting on the pessimistic hat, getting * to work under those conditions w/ the # of ports (48) you're discussing.. I think is probably your biggest headache. I wrote 4 other paragraphs about what I think, and deleted them. Interesting, let me know where you go with this. -m - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Greg Boehnlein wrote: On Wed, 15 Dec 2004, Matt Klein wrote: who said anything about a computer? :) computer, $$extra on both. may be less on the pm3 side due to resource needs. In the scenario I envision this being used in, there is no computer. The PM3 runs (On it's x86 w/ 4 or 16 megs of ram) a stripped down, embedded version of Linux + Asterisk. With a TE405P you need a PC to house the cards in. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 and analog cards FXS in one box.
Hi all ! I want use 8 FXS ports maybe more with E1 card from Digium or Sangoma. All together in one Asterisk box. I heard that are some problems with mix analog and digital card in this same machine. Is that true ?? Does someone has this instalation ?? Thanks Andrzej ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
they've mentioned interest in making it a channel bank, really, FXS/FXO to SIP or IAX or another protocol, delivered via tcp/ip, and your input would be interesting regarding the hardware capabilities of the boxes. - I believe there are more instances of the abridgment of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations. - James Madison On Wed, 15 Dec 2004, Bob Knight wrote: On Wed, 15 Dec 2004, Matt Klein wrote: 3) good luck getting the firmware source is the firmware source freely available, -- I've been asked by others. All the other (excellent, thought provoking) conversation aside, Jake Messenger from Portmasters.com has been granted a license by Lucent for ComOS. http://www.portmasters.com/pipermail/comos/2004-August/41.html That contains a link to the license the source is under. It isn't free as in GNU, but I don't think that really matters much. I had to give up following this list too closely, because it just sucks up too much time. But I did just stumble onto this thread about portmasters. I worked at Livingston and wrote the drivers on the portmasters. That source code is easy to find and even compiles on a linux box these days (we used to use SunOS). If you come up with anything interesting to do with the boxes, please let me know I may be able to help. Contact me off list is best. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP bad voice quality
On Thu, Dec 16, 2004 at 11:35:22AM +0530, Ashish Shinde spake thusly: How can I solve this problem of voice quality? Can a better implementation of jitterbuffer with packet loss concealment help? If so how do I get the newer implementation. I would really like to help out in the development of the new jitterbuffer if it has not yet been implemented. You are going over the open Internet which does not have QoS. Unfortunately it is unlikely that you will be able to get rid of that 1% of packet loss which will cause the occasional break in your audio. Perfect audio is just not possible over an imperfect transmission medium using current technology. Packet loss concealment might help to some degree but the audio quality will still suffer, although perhaps it won't be as obvious. Unfortunately such technology is not currently available in asterisk. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpIiVRvGWcU2.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
Matt Klein wrote: they've mentioned interest in making it a channel bank, really, FXS/FXO to SIP or IAX or another protocol, delivered via tcp/ip, and your input would be interesting regarding the hardware capabilities of the boxes. Please strongly consider having it do IAX. It solves a lot of problems. I wish there were more hardware out there that spoke it natively. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls arent handled by asterisk - destruction of call
Hello, Im trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isnt answered by my server when I try calling the number that I registered at my SIP provider. Ive registered with register = John.Doe:MyPass:[EMAIL PROTECTED]/1000 in sip.conf and if I use sip debug I can see the call is coming in but then nothing more happens (see debug output below). Also get these error messages:Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msWARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Can you guys help me?Thanks :) Sip.conf:[general] context=demo [my-sip-provider] type=peer fromuser=MyUser secret=MyPass fromdomain=my-sip-provider context=demo extensions.conf: [demo] ; ; All the stuff in the demo ; exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,n,Answer ; Answer the line exten = s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds and so on Thats all I have have I missed something? Debug output from call: 192.1.1.1=my server 0123456789=my number at SIP-provider 99=the number Im calling from 213.132.103.213, 212.112.162.50=my SIP providers IPs == Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:213.132.103.213:5060;transport=UDP;lr=true Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b Record-Route: sip:[EMAIL PROTECTED];ftag=2EBE3E60-1646;lr Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: sip:[EMAIL PROTECTED];tag=2EBE3E60-1646 To: sip:[EMAIL PROTECTED] Date: Wed, 15 Dec 2004 10:10:11 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1458717796-1303908825-2510524757-306778262 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 9 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1103105411 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 288 v=0 o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22 s=SIP Call c=IN IP4 212.112.162.22 t=0 0 m=audio 16842 RTP/AVP 18 0 101 c=IN IP4 212.112.162.22 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 24 headers, 12 lines Using latest request as basis request Sending to 213.132.103.213 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 212.112.162.22:16842 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found peer 'wx3.se' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: sip:[EMAIL PROTECTED];tag=2EBE3E60-1646 To: sip:[EMAIL PROTECTED];tag=as3c0db481 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm="asterisk", nonce="59e60c89" Content-Length: 0 to 213.132.103.213:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 User-Agent: sapphire/1.6.2.0253 Max-Forwards: 70 Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa69cda50ac3600b To: sip:[EMAIL PROTECTED];tag=as3c0db481 From: sip:[EMAIL PROTECTED];tag=2EBE3E60-1646 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP bad voice quality
Hi, Thanks for the reply and sorry for the multiple mails. My mail client kept on giving errors in sending and this was kinda urgent. I was just wondering if the G711 codec with the PLC algorithm might help me out with the 1% packet loss. Guess other people might also be facing similar problems, how do they cope up with it? Thanks and regards, - Ashish On Thu, 16 Dec 2004 00:30:28 -0800, Tracy R Reed [EMAIL PROTECTED] wrote: On Thu, Dec 16, 2004 at 11:35:22AM +0530, Ashish Shinde spake thusly: How can I solve this problem of voice quality? Can a better implementation of jitterbuffer with packet loss concealment help? If so how do I get the newer implementation. I would really like to help out in the development of the new jitterbuffer if it has not yet been implemented. You are going over the open Internet which does not have QoS. Unfortunately it is unlikely that you will be able to get rid of that 1% of packet loss which will cause the occasional break in your audio. Perfect audio is just not possible over an imperfect transmission medium using current technology. Packet loss concealment might help to some degree but the audio quality will still suffer, although perhaps it won't be as obvious. Unfortunately such technology is not currently available in asterisk. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My Boss wants background music!!!!
Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server it might be at the maximum 570Kbs if it has to send it individually to each telephone. The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone. Please, is there a way to get this done, otherwise I have to say goodbye to Asterisk (unless my boss gives in). Thank you all Willy ONS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kewlstart - explanation of this method, please ?
Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support it.. but the discussion that I found below in http://massis.lcs.mit.edu/archives/back.issues/recent.single.issues/V23_%23392, seems to suggest kewlstart already having such a feature. Where can we learn more about what kewlstart does and what / how it does it ? -samudra In article [EMAIL PROTECTED], [EMAIL PROTECTED] says: [EMAIL PROTECTED] (Robert Bonomi) writes: In article [EMAIL PROTECTED], Kyler Laird [EMAIL PROTECTED] wrote: I'm trying to set up a home PBX and I decided to just take a crack at getting kewlstart/calling party control/disconnect supervision on my home line. I called Verizon and got bounced around until I hit someone with 31 years of experience who had never heard of such a thing. I was told that Verizon certainly doesn't offer it. I suspect that someone in Verizon knows how to provision the switch and can twiddle a few bits to give it to me. Is that reasonable? How do I find that person? No it is _not_ reasonable. Not for a _residential_ POTS phone line. If you want to pay for a 'commercial rates' _trunk_ line, Then you can start talking about things like wink start vs ground start vs loop start, EM vs TR, MF vs DTMF signalling, etc., etc., ad nauseum. FWIW: kewlstart isn't a telco line type like a loopstart or groundstart trunk line. Its a special mode of the Asterisk soft PBX system that takes a normal loopstart line (ie. a POTS line) and watches for a certain event on it to handle line drops (ie. remote disconnect detection) better than normal loopstart signalling. (ie. a posting on the Asterisk users archives from the main author kewlstart is what we call loopstart with battery drop. this is also known as far end disconnect supervision to some people. Basically when the switch hangs up on you, it drops battery for a fraction of a second to signal that you've been hung up on. As such, you won't find any telco offering it, because its a special mode that Asterisk has for its FXO cards on a plain old loopstart telephone line. Its not surprising at all that nobody at any telco has heard of it, and the OP is barking up the wrong tree for nothing. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.804 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk on FC3
Hello, Since FC3 has been a very recent release I was just wondering if there are issues related to asterisk installation on FC3. Thanks Varun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk on FC3
I have Asterisk (the yesterday CVS) installed on FC3. No issues so far. On Thursday 16 December 2004 12:48, [EMAIL PROTECTED] wrote: Hello, Since FC3 has been a very recent release I was just wondering if there are issues related to asterisk installation on FC3. Thanks Varun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Teodor Georgiev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g711 ulaw vs alaw
Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are similar, they sound the same and that it doesn't matter which you use. Could someone knowledgable please enlightmen me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g711 ulaw vs alaw
Partly is is down to the fact that G.711u (mu-law) is primarily used in the USA and G.711a (a-law) is used in Europe. Like you, I am not sure if the exact differences - they have the same bitrate and audio, although there are minor differences in the format. Peter -Original Message- From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: 16 December 2004 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] g711 ulaw vs alaw Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are similar, they sound the same and that it doesn't matter which you use. Could someone knowledgable please enlightmen me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring an active call
Hello, Would it be possible to monitor an extension in asterisk real time (not record and than monitor). To call an extension on asterisk and be able to monitor specific extensions, by punching in that extension number, maybe a password too (for training purpose). The calls are not in conference group, just a regular call from extension to outside number. If yes how can we set it up? Thanks in advance Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Boss wants background music!!!!
On Thu, 2004-12-16 at 04:48, Satchid wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server it might be at the maximum 570Kbs if it has to send it individually to each telephone. The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone. Most hardware phone systems will do this, and its circuitry is related to station paging and hands free autoanswer intercom. I was wondering about this feature myself, but more interest in station paging and HFAI. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Boss wants background music!!!!
Satchid wrote: Dear Members, As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server it might be at the maximum 570Kbs if it has to send it individually to each telephone. The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone. You can do this with a multiline phone by setting up an extension like this: exten = 201,1,SetMusicOnHold(emp) exten = 201,2,Answer() exten = 201,3,MusicOnHold() exten = 201,4,Hangup Then be sure you have the emp music class setup in musiconhold.conf: emp = quietmp3nb:/var/lib/asterisk/mohmp3/emp,-z This works fine. Users use one line to dial in to this and let it run on speaker phone. When a call comes in, it is on another line and can be picked up. When they hang up, they resume the music line. -- respectfully, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 190 Call Completion
I cannot get Call Completion to work on Snom 190, does anybody have it working? I have set CC=yes, and when I call another phone the phone display CC at the softkey, when I press it I get the options OK/Cancel. I press OK and then the display shows CC. This will not change until I call the phone I have on CC, then it disappears what am I missing? Thank you Thorben Is nobody using this facility? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring an active call
Hello, Would it be possible to monitor an extension in asterisk real time (not record and than monitor). To call an extension on asterisk and be able to monitor specific extensions, by punching in that extension number, maybe a password too (for training purpose). The calls are not in conference group, just a regular call from extension to outside number. If yes how can we set it up? Thanks in advance Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logging codec in cdr?
hi Is it possible to log the codec used in CDR? Today, I have an AGI script logging the ipaddr of the sip client to the userfield. how can I find the current codec as reported on the console: -- Format for call is g726 roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virtual Modems
Hi, Miguel Ruiz Velasco Sobrino schrieb: why do you want to relay a modem over an VoIP network? isn't it no-sense to warp digital data inside analog signals to be warped over digital data?. I believe that is the point. The really wicked implementation would be to have a codec converter that would compress the audio data by reducing it to the actual data transferred. A probably nicer implementation would be app_modem that would emulate a modem and call some external application with stdin/stdout connected to the caller. Simon (placing it on line 1538 of the todo list) signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g711 ulaw vs alaw
Whisker, Peter schrieb: Partly is is down to the fact that G.711u (mu-law) is primarily used in the USA and G.711a (a-law) is used in Europe. Like you, I am not sure if the exact differences - they have the same bitrate and audio, although there are minor differences in the format. Hi, it is just a small difference in how to interpret one bit (signed value vs. unsigned value with offset). Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automated callback with .call file
Hello, I am attempting to write a script to launch a callback based on a dial-in service. I have created this call file: --- channel: IAX2/[EMAIL PROTECTED]/011_valid_number maxretries: 3 retrytime: 5 waittime: 5 context: dialtone extension: 912125551212 priority: 1 --- Where I first attempt to dial the callback user (channel) and then connect the call to the number they dialed (extension, which is valid for connecting via the dialtone context), I get the following error: --- -- Attempting call on IAX2/[EMAIL PROTECTED]/011_valid_number for [EMAIL PROTECTED]:1 (Retry 1) -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/voipjet/1' Dec 16 12:41:17 WARNING[20464]: app_queue.c:340 changethread: Can't change device with no technology! Dec 16 12:41:17 NOTICE[20464]: pbx_spool.c:234 attempt_thread: Call failed to go through, reason 0 --- If I change the voipjet to an internal SIP extension (ie - SIP/1000) it works fine. How do I get this working? Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Boss wants background music!!!!
At 04:48 AM 12/16/04, you wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. If you don't mind my asking, what application would require this feature? Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g711 ulaw vs alaw
On Thu, 16 Dec 2004, Roger Schreiter wrote: Whisker, Peter schrieb: Partly is is down to the fact that G.711u (mu-law) is primarily used in the USA and G.711a (a-law) is used in Europe. Like you, I am not sure if the exact differences - they have the same bitrate and audio, although there are minor differences in the format. it is just a small difference in how to interpret one bit (signed value vs. unsigned value with offset). I think there is a bit more difference. The byte code of ulaw is a monotonic function of the amplitude whereas in alaw the code is xor:ed with a bit mask of 0x55. Also, the quatization function is different, though the same general idea is used. If the difference is important to you you should read the standard itself. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Pager Subject?
Hi,
I have set emailsubject in voicemail.conf as follows:
emailsubject=New Voicemail from ${VM_CALLERID} in Mailbox ${VM_MAILBOX}
This works fine, but the pager e-mails come through with New VM. I
would like the pager e-mail to be the same subject as above as I get
this as an SMS to my mobile phone.
I thought this would be config'able but I can't find any mention of it
on the wiki or in any examples. I've tried setting pagersubject (a
guess) and this didn't do anything.
Is this possible or is the New VM subject for pagers hard coded in the
source code?
An alternative if this isn't possible is to remove the pager and just
send the e-mail version to my phone - can you send a voicemail
notification to multiple e-mail addresses or do I have to create an
alias on my mail server?
Thanks!
--ian
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[Asterisk-Users] Making sip show channels show sane results with sipfriends from mysql?
hi using sipfriends from mysql from asterisk 1.0 branch, how can I make asterisk show the true channel's current codec with SIP SHOW CHANNELS? This does not seem to work, and bkw_ said sipfriends from mysql didn't have that info at all. For what it may seem, asterisk uses G.726 as told, giving me a -- Format for call is g726 at the start of the call, but in SIP SHOW CHANNELS all these turn up as ALAW calls, although SIP DEBUG looks like it's really running G.726. It'd be really nice to see some real data here... roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec preference?
hi With a setup like this SIP client - asterisk1 - IAX2 asterisk2 - PSTN will asterisk1 and asterisk2 choose codec based on the codec used by SIP client - asterisk, or only by preference in sip.conf/iax.conf? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g711 ulaw vs alaw
I think there is a bit more difference. The byte code of ulaw is a monotonic function of the amplitude whereas in alaw the code is xor:ed with a bit mask of 0x55. Wow! Encryption! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 - Stop Message Waiting Indication
Hi, I have my Sipura SPA-3000 setup with Asterisk as follows: [spa3k_line1] type=friend context=home secret=PASSWORD host=dynamic dtmfmode=rfc2833 dissallow=all allow=ulaw When an incoming call comes in, I have a Zap interface in Asterisk which just does a Wait,15 then answers with voicemail. The SPA-3000 detects the PSTN call and makes Line 1 ring - so I can answer the phone if i'm around - if not, Asterisk gets to the end of the 15 seconds and answers the call. However - I have a strange problem - after a person leaves a voicemail message, I get a stuttered dial tone on the phone attached to Line 1 of the SPA-3000 and it does 1 ring every now and again. How can it be doing this if I dont have a mailbox= line in the asterisk config? Any ideas how to stop the SPA-3000 detecting a message is waiting? For now, I think i've got rid of it by disabling MWI and VMWI on the SPA-3000 config but it would be nice to do it in Asterisk so my spa3k config is as generic as possible. Thanks --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QOS Device?
On December 15, 2004 11:40 pm, Me wrote: Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? Put iproute2 and tc on the firewall. Limit the traffic out of the firewall to the T1 router to 1500kbps. There are tons of shaping scripts out there but I prefer something I rolled together and use myself: http://www.mixdown.ca/~andrew/dump/rc.tc. I don't profess to be a traffic shaping guru and if anyone has any suggestions on how to make it better I'd be grateful but it seems to work very well for me. I can completely saturate my ADSL uplink (800kbps) without really bad degradation in my outgoing audio (there is some but it's not bad according to the other side). I can't change the T1 router to something that supports QOS because it has certain redundant features with an ISDN line which are needed. You could always use a Sangoma A101u and a CAPI card in a Linux box. :-) The HDLC features in the Digium T100P are still being ironed out, IIRC. Sangoma's been doing it for literally years if not coming up to a decade yet. I've got a friend who's been using an older (DSU/CSU-less) version of their T1 card for at least that long. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Termination
On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote: One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the phone, and one that works well with Asterisk. You've gotta be kidding, VOIP providers are trying to regulate who you can call? Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over SIP, IMO it's just better. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QOS Device?
what are you running on Fedora Core to shape the traffic? Traffic Control tc is included in the 2.4 kernel and forward. See http://lartc.org/. Basically, I have a script that is setup as a service to set up the bridge and the traffic control queues. let's say that you have VPN site to site tunnels from the FW behind the QoS machines towards a branch office and that some of the traffic in the Tunnel has higher priority then other traffic. The QoS device sees it all as encrypted traffic and can't help there. What would you suggest? If you want to shape VPN traffic, then you would need to place the QoS behind the VPN box. So long as you can route _all_ of your WAN traffic through QoS, it will be effective. Our VPN traffic is all considered 'bulk' traffic so it isn't a concern of our setup. Encrypted traffic is still a pain though. With Citrix for example, all of our users are hitting the Metaframe server which has all traffic encrypted with SSL all the way back to the client. So... I'm unable to separate out Citrix printer traffic from interactive traffic. I just have to look at source / destination (IP of our colocation facility) to determine priority. We were able to come up with kind of a workaround though. We put in a print server at colo instead of printing directly from the clients. So this way the print server connects over the VPN to send a print job to a printer. That print job then becomes bulk traffic. Pretty neat trick IMO. ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxs causing constant CPU spikes
On December 15, 2004 10:54 pm, Eric Bishop wrote: I'm not sure if it is the CPU spikes or not but there definately is a high level of flakiness with the card. ie very low gain levels and often for no reason will refuse to accept calls until the driver is reloaded. Honestly at this time I am recommedning all my customers with Analog lines to stick with their traditional PBX and just attach an FXS VoIP gateway to it. This is unfortunately what I do, too -- T1+channel bank in lieu of a TDM400P -- I know for *certain* that Digium's working on it, but they are being awful quiet about it. As far as CPU spikes being the culprit of this, I think it's normal operation -- I have a Digiumless box (A101u+S518) that has similar CPU spikes but zttool is showing that the timing is spot-on, and audio is perfect. I am going to assume that the CPU spikes are likely in the zaptel driver (which the A101u hooks in to for Asterisk support) and not in the wctdm driver. From what I understand the crackly audio on FXS and problematic pickup/hangup on FXO is due more to static and/or noise issues, not driver issues, but I very well could be mistaken. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk -- Nuera Orca
Hi, I am wondering if it is possible to interconnect two Nuera ORCA GX gateways via Asterisk. Nuera ORCA gateways can speak only MGCP and usually they are connected via MGCP to Nuera SSC (Softswitch) that speaks MGCP + SIP. Is there someone, who has tried it? Any points are welcome :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect line is busy with Zap?
Hi,
I have an FXO card connected to my phone line which works in Asterisk as
Zap/1.
Is there any way of detecting whether something else is on the line
before picking up on this channel?
For example, I dont want to pick up and dial out on the line if someone
is on it using another phone (which is connected directly to the line,
rather than through Asterisk).
Also, when an incoming call comes in, i've got this:
[incoming_pstn]
exten = s,1,NoOp(${CALLERID} calling on Zap/1)
exten = s,2,Wait,15
exten = s,3,Answer
exten = s,4,Voicemail(su1)
exten = s,5,Hangup
I really want to wrap the Answer, Voicemail and Hangup commands in some
kind of if statement so if Asterisk didn't detect that the call had been
answered for any reason, it wouldn't answer the call if the line was
busy - otherwise it would click in and play the voicemail greeting over
the call which someone is conducting from a non-asterisk phone on the
line.
My zapata.conf looks like this:
[channels]
context=incoming_pstn
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1.0
txgain=0.0
channel = 1
Thanks!
--ian
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Re: [Asterisk-Users] VoIP Termination
On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote: One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the phone, and one that works well with Asterisk. You've gotta be kidding, VOIP providers are trying to regulate who you can call? Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over SIP, IMO it's just better. Think what he might be suggesting is that he signed up for an unlimited residential plan, and those policies try to minimize business use (or high volume) calling. I'd be real curious how the itsp 'actually' attempts to enforce that given that a fair number can't even deal with simple issues. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxs causing constant CPU spikes
Anybody know what the 'real' differences are between a PC's pci/interrupt implementation and that of the older Mac implementation (that has been frequently recommended)? Why is it there seems to be issues with PCs but not Macs, both running linux? Rich I'm not sure if it is the CPU spikes or not but there definately is a high level of flakiness with the card. ie very low gain levels and often for no reason will refuse to accept calls until the driver is reloaded. Honestly at this time I am recommedning all my customers with Analog lines to stick with their traditional PBX and just attach an FXS VoIP gateway to it. On Wed, 15 Dec 2004 22:03:43 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On December 15, 2004 09:27 pm, Michael Welter wrote: Yes? Is there a workaround, or do I tell my customer to go find something else? Are the CPU spikes causing you trouble or did you just notice them when poking around? I see CPU spikes but don't have any issues at all (noise or otherwise). -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g711 ulaw vs alaw
On Thu, 16 Dec 2004, Roy Sigurd Karlsbakk wrote: I think there is a bit more difference. The byte code of ulaw is a monotonic function of the amplitude whereas in alaw the code is xor:ed with a bit mask of 0x55. Wow! Encryption! I think it is done to keep the number of 1 and 0 on the line roughly equal. For some links this was/is desireable. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
Polycom IP300 are very in-expensive and have multiple lines. On Thu, 2004-12-16 at 13:49 +0100, Satchid wrote: What ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
At 07:49 AM 12/16/04, you wrote: Thank you, My boss believes that people are more happy when soft music playing in the background. The volume has to be low or even of when the phone rings. If this is coupled to the *, then the volume can automatically switch of or switch low. Therefore the volume can be set as high as the user wants because when the phone rings it switches of. With the existing very rudimentary PA system the volume has to be low all the time because of not being coupled. Thanks for your response, now I have a better understanding. I like the 2 line telephone system idea, but it will probably be much more expensive. I was planning the Grandstream 102 phones. What phones would you propose me to use? Willy The Grandstream phones are priced well. we currently use the bt100's. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg - Cirelle Enterprises Sent: Thursday, December 16, 2004 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] My Boss wants background music At 04:48 AM 12/16/04, you wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. If you don't mind my asking, what application would require this feature? Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reporting Errors Mysql
When I Dial(sip/[EMAIL PROTECTED]) I get a SIP responce 404 not found Now.. is there any way I can use this error (string) in an agi script? or log it somwhere? I'd like to have some good error reporting for calls and have them stored in mysql. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] send # with transfer enabled
Every so often we need to send the # dtmf tones but * interprets that as the initiation of a transfer. The best solution I've found so far is outlined at: http://lists.digium.com/pipermail/asterisk-users/2004-March/039501.html This disabled transfer for a call. I take this to mean that there is no way to send dtmf for # with transfer enabled? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk on FC3
Teodor Georgiev wrote: I have Asterisk (the yesterday CVS) installed on FC3. No issues so far. On Thursday 16 December 2004 12:48, [EMAIL PROTECTED] wrote: Hello, Since FC3 has been a very recent release I was just wondering if there are issues related to asterisk installation on FC3. Thanks Varun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have installed Asterisk on FC3 with all the current patches on an AMD Opteron 64 bit platform - this weekend I will transfer all of my working configs and cards from my old Asterisk box to the new box - I will report any issues that may arise. BTW - the AMD is one fast box - compiling Asterisk and all associated components from cvs/scratch took about 2 minutes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect line is busy with Zap?
Inline...
I have an FXO card connected to my phone line which works in Asterisk as
Zap/1.
Is there any way of detecting whether something else is on the line
before picking up on this channel?
The simple answer: No. The long answer: the chipset used for the tdm fxo
modules does have the capability to sense that (along with other things),
however the driver code does not attempt to make use of such data.
For example, I dont want to pick up and dial out on the line if someone
is on it using another phone (which is connected directly to the line,
rather than through Asterisk).
Also, when an incoming call comes in, i've got this:
[incoming_pstn]
exten = s,1,NoOp(${CALLERID} calling on Zap/1)
exten = s,2,Wait,15
exten = s,3,Answer
exten = s,4,Voicemail(su1)
exten = s,5,Hangup
I really want to wrap the Answer, Voicemail and Hangup commands in some
kind of if statement so if Asterisk didn't detect that the call had been
answered for any reason, it wouldn't answer the call if the line was
busy - otherwise it would click in and play the voicemail greeting over
the call which someone is conducting from a non-asterisk phone on the
line.
I'm not sure I understand all the double-negatives in that statement,
but try a couple answers to see if it fits with what you're trying
do do.
I have one incoming pstn line that has both * and analog house phones
connected to the same pstn line. I use:
[inbound-home]
exten = s,1,Dial(${PHONE3}${PHONE4},20)
exten = s,103,Hangup
where the 20 second timeout is longer then our old answering machine
uses (that my spouse loves). If she picks up the analog phone (non-*),
* detects the fact that ringing disappeared and ignores the call.
If no one answers, * still ignores the call after 20 seconds. If
you want * to answer after the 20 seconds (assuming no one picked
up the call from any source), then something like add:
[inbound-home]
exten = s,1,Dial(${PHONE3}${PHONE4},20)
exten = s,2,Voicemail(u3000)
exten = s,102,Voicemail(b3000)
exten = s,103,Hangup
If one of those suggestions does not fit your requirements, then
you'll likely have to insert * in the middle of all calls. E.g, receive
the call on a fxo port, and route that call to a fxs port, leaving
asterisk in the middle with whatever logic you want to apply. That
would require something like the tdm card with both fxo and fxs
modules, or, spa-3000 (for substantially less money for one line).
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Re: [Asterisk-Users] Detect line is busy with Zap?
Ian Chilton [EMAIL PROTECTED] wrote: [...] Is there any way of detecting whether something else is on the line before picking up on this channel? No, but you could insert a privacy adaptor device to prevent the FX100P from attempting to dial out. -- My mother protected me from the world and my father threatened me with it. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec order in SIP doesn't work
Seems this is only partially. This is from latest CVS -r v1-0 If specifying a peer/friend in sip.conf without any codec prefs, I get this Username : 112 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (g726|g729|gsm|alaw) but if not specifying it in sip.conf, and rather using sipfriends from mysql Username : 112 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (none) so there's something missing here, it seems More testing follows: http://pastebin.ca/raw/3036 shows a SIP DEBUG when a client (supporting G.726, alaw and ulaw) connects to server supporting the following (from sip.conf) disallow=all allow=gsm,g726,g729,alaw for connections from some x-pro phones, running GSM codec, SIP SHOW PEERS shows correct codec. however, for those like the SIP DEBUG pasted above, the SIP SHOW PEERS shows ALAW. Please, someone, help me out here. roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] unsubscribe
Bart de Wild [EMAIL PROTECTED] wrote: unsubscribe No. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queueueueuueue position
Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ;Announcement to play to the Agent answering strategy=ringall;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them an estimated hold time queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) join-empty=yes leavewhenempty=no member=Agent/1000 When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] send # with transfer enabled
There are many bugs that track this feature request. Several of us have been trying to get it into CVS at various times over the last year(Mark doesn't like how any of the patches have been done, so we keep having to patch for new versions over and over again). It is configurable in the features.conf file. take a look at these bugs associated with this feature request(not sure if the patch in the first one works with current CVS): http://bugs.digium.com/bug_view_page.php?bug_id=0002010 http://bugs.digium.com/bug_view_page.php?bug_id=885 Good luck, MATT--- -Original Message- From: Michael George [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 8:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] send # with transfer enabled Every so often we need to send the # dtmf tones but * interprets that as the initiation of a transfer. The best solution I've found so far is outlined at: http://lists.digium.com/pipermail/asterisk-users/2004-March/039501.html This disabled transfer for a call. I take this to mean that there is no way to send dtmf for # with transfer enabled? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2-NA Screenshot
Of the 15 or so screens in the PAP2-NA web interface, which did u want to see? =) -Matthew - Original Message - From: M. Ehsanul Karim [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 8:19 PM Subject: [Asterisk-Users] Linksys PAP2-NA Screenshot Hello, Will it be possible for any one to send me a screenshot of the new Linksys PAP2-NA web panel. I would appreciate it very much. Thanks, Ehsanul Karim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Caller ID Name PRI NI2.
On Wed, Dec 15, 2004 at 10:02:46PM -0700, Chris Modesitt wrote: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 852/0x354) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 96] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 22 ] [6c 0c 21 83 38 30 31 37 38 37 34 39 30 36] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '8017874906' ] [70 0b a1 38 30 31 34 33 37 37 38 36 30] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8014377860' ] -- Making new call for cr 852 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) You are definitely not getting calling name over facility information element from your telco. I do not see calling name information anywhere in that dump. Unless that's not a complete dump, or there's some other problem, I'd talk to your telco about it. Matthew Fredrickson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk + freeradius
Hmmm, I can give you some help. I don't use it FreeRadius but I do write programs and make modifications with. First, (warning these are very harsh, even tyrannical questions.) 1. How much have you looked in the wiki's for configuration stuff? 1.a Did you install and configure the freeradius? 2. Do you have a radius.conf in your /etc/asterisk directory? 3. What are you expecting from RADIUS? My guess is that you want CDRs. If nothing else I can lead you in the wrong direction and someone else will answer your question because the level of detail in your question will rise. Race The Tyrant Van der Decken In the Land of the Blind the One-Eyed man is Elvis. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Gabriel Drach Sent: 15 December 2004 17:56 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk + freeradius Hi i am lost on how to configure asterisk + freeradius. i am not sure how app_radius.so work. if somebody can give some hints... Bye Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple IAX client behind a NAT
Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, it goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 (SIP) hold problems
Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the other end a diff type of trunk ie: 7960sip -- asterisk -- IAX2 -- PRI 7960sip -- asterisk -- SER -- SIP proxy Anyone have a clue? The 7960 has the latest firmware, 7.3 or something. Could this be a (the?) problem? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX client behind a NAT
I wonder if you can put can reinvite=yes in the iax2.conf file like we use in our sip.conf file to do what you are requesting. I believe it should tell the phones to do what you wish On Thu, 2004-12-16 at 17:40 +0200, CuPoTKa wrote: Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, it goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: iax.cc hosts - want to do some traceroutes before buying
Sorry about this, but do any users have more detailed iax.cc information? Will they do trunking? What are the hosts that I will be logging into? I want to make sure that they will work well for me, and I would like to do some traceroutes to make sure that they are close! Thanks. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
The first example wasn't even touching SER.. 7960sip -- asterisk -- IAX2 -- PRI :/ -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) hold problems Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the other end a diff type of trunk ie: 7960sip -- asterisk -- IAX2 -- PRI 7960sip -- asterisk -- SER -- SIP proxy Anyone have a clue? The 7960 has the latest firmware, 7.3 or something. Could this be a (the?) problem? Thanks! I'm not aware of any issues. One remote internet based with g729 and nat, another with g711, and several local. If its happening here, no one knows about it. We're not using SER though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX client behind a NAT
I tried notransfer=yes/no (like reinvite for the sip.conf), but the calls still go throuht the asterisk. Justin Carlson wrote: I wonder if you can put can reinvite=yes in the iax2.conf file like we use in our sip.conf file to do what you are requesting. I believe it should tell the phones to do what you wish On Thu, 2004-12-16 at 17:40 +0200, CuPoTKa wrote: Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, it goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Alex Massover n:Massover;Alex org:Gamcom Ltd., Gamcall Ltd. email;internet:[EMAIL PROTECTED] title:Chief Technology Officer tel;work:+(972)-9-8653633, sip:[EMAIL PROTECTED] version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to tell Who's Online?
if I'm missing something obvious, but I couldn't find any console command to show users online. sip show peers iax2 show peers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to tell Who's Online?
On Thu, 2004-12-16 at 17:41 +0100, Wilson Pickett wrote: if I'm missing something obvious, but I couldn't find any console command to show users online. sip show peers iax2 show peers Thank you, Do you know, if an IAXy device (or anything else speaking IAX2) disappears, how long will it be (minutes, hours?) before Asterisk notices they are offline, and iax2 show peers will reflect the change of online status? Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Phone Suggestions
I have two UIP200s at home and they work great for me...they do have superior sound qualit like they other user siad. Make sure you ask their tech support for the latest "test" release firmware which was 4.62h when I got it a few weeks ago.Default firmware loaded on the phone has several compatibilityissues with *.The new firmwaresupports phone book, call log, STUN (which you can use for NAT traversal) and fix all known * problems. Don Me [EMAIL PROTECTED] wrote: I have one but never was able to get it to ring with or without a NAT in front of it. Calls out worked fine. There seemed to be only one person supporting this product at Uniden, he was very nice but after 4 or 5 calls I just gave up. The phone now collects dust on one of the desk in the office. Also, I was told several times that the phone will not work at all behind a NAT. I tried it at the office where there is no NAT in between the phone and the * box but still could not get it to ring. Todd --Start Your Own ISP!http://www.YourOwnISP.com - Original Message - From: Kevin Curtis To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 15, 2004 10:53 PM Subject: Re: [Asterisk-Users] VOIP Phone Suggestions I would recommend Uniden UIP200 phones. Great sound quality with inbuilt phone book, call logs etc works great with asterisk. I recently purchased from www.qualvoip.com (they also provided me sample configuration files for asterisk). Kevin - Original Message - From: Shawn Dillon To: [EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 8:39 PM Subject: [Asterisk-Users] VOIP Phone Suggestions We are in the final stage of a rollout of Asterisk in our company. We had some Polycom IP 600 , a Snom 220 , a Grandstream 102 and recently a Sayson 480i phone. I am interested in anyones opinions in the phone they suggest to implement. I must admit I am a little partial to the Sayson 480i , but if there are convincing arguments with regards to other models I would like to hear them. If anyone has had more experience with the Sayson please let me know. There is a company in Vancouver that deals in them , call NetVoice. As a newbie in the market , they ( George) gave great service and advice. Even called me to see how the Snom 220 was working out ( Great customer service!!). Anyways , your feedback is appreciated. Shawn ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? All your favorites on one personal page Try My Yahoo!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] Cisco 7960 (SIP) hold problems
ala cisco 7960 -Original Message- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck. sip.conf [107] host=dynamic type=friend context=default username=107 secret=blahblah mailbox=107 canreinvite=no disallow=all allow=all -- -sipMACADDRESS.cnf- image_version: P0S3-07-3-00 line1_name: 107 # Line 1 Registration Authentication line1_authname: 107 # Line 1 Registration Password line1_password: elblahblah --snip-- ### New Parameters added in Release 2.0 ### # All user_parameters have been removed # Phone Label (Text desired to be displayed in upper right corner) phone_label: Matt S 107 ; Has no effect on SIP messaging # Line 1 Display Name (Display name to use for SIP messaging) line1_displayname: Matt S # Line 2 Display Name (Display name to use for SIP messaging) line2_displayname: ### New Parameters added in Release 3.0 ## # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: SIP Phone ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: blahblahblah ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none - sipdefault.cnf # Image Version image_version: P0S3-07-3-00 # Proxy Server # Note: I put the proxy server information in the individual conf files # for each machine, since each box has different configs. You could, of course, # put all of them here in the Default file... proxy1_address: 192.168.1.17 #proxy2_address: 192.168.117.4 # Proxy Server Port (default - 5061) #proxy1_port:5060 # Emergency Proxy info proxy_emergency: 192.168.1.17 proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: 192.168.1.17 proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: 192.168.1.17 outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5061 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 0 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 120 # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Enable VAD (0-disable (default), 1-enable) enable_vad: 0 # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: 1 ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 0 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 11 sip_invite_retx: 6 ; Default 7 timer_invite_expires: 180; Default 180 sec # Setting for Message speeddial to UOne box messages_uri: 8500 #* Release 2 new config parameters ** # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ./ # Time Server sntp_mode: directedbroadcast sntp_server: 17.254.0.49 time_zone: CST dst_offset: 1 dst_start_month: April dst_start_day: dst_start_day_of_week: Sun dst_start_week_of_month: 1 dst_start_time: 02 dst_stop_month: Oct dst_stop_day: dst_stop_day_of_week: Sunday dst_stop_week_of_month: 8 dst_stop_time: 2 dst_auto_adjust: 1 # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: 1 ; Default 1 (Call Waiting enabled) #
Re: [Asterisk-Users] How to tell Who's Online?
if I'm missing something obvious, but I couldn't find any console command to show users online. sip show peers iax2 show peers Thank you, Do you know, if an IAXy device (or anything else speaking IAX2) disappears, how long will it be (minutes, hours?) before Asterisk notices they are offline, and iax2 show peers will reflect the change of online status? Take a close look at parameters documented in /usr/src/asterisk/configs/sip.conf.sample and you'll see something like: ;qualify=1000 ; Consider it down if it's 1 second to reply That statement essentially tries to contact a sip phone every second. If it can't each the phone, you will see unreachable messages on the CLI and in the 'sip show peers'. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced Ring All Hunt Group
Here is an idea to try. Maybe someone else has a cleaner solution.
exten =
9043442342,1,Dial(local/[EMAIL PROTECTED]local/[EMAIL PROTECTED]local/[EMAIL
PROTECTED]l
ocal/[EMAIL PROTECTED],,20)
exten = 9043442342,2,Voicemail(u102)
[rollover]
exten = _10X,1,Dial(SIP/10${EXTEN:2},,21)
exten = _10X,102,Dial(SIP/20${EXTEN:2},,21)
exten = _10X,203,Dial(SIP/30${EXTEN:2},,21)
exten = _10X,304,Busy
See if that works for you.
B. J.
From: voipbuilder [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 15, 2004 16:56
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Advanced Ring All Hunt Group
Hello Everyone,
I need to setup a dialplan where if a incoming call is rec'd to a number,
Asterisk needs to dial several SIP extensions at the same time. The SIP
extensions are for Cisco 7960s and each have multiple line appearnces.
For example,
exten = 9043442342,1,DIAL(SIP/102SIP/103SIP/104SIP/105,,20)
exten = 9043442342,1,Voicemail(u102)
The issue I have is that I need each user of these extensions to have
multiple line appearances (roll over lines). In a traditional PBX,
usually this is accomplished by setting up a roll over lines...
i.e my extension is 100, my roll over extension is 200, and next roll over
extension is 300. So if i am on my first line, the next call will roll over
to 200.
I have this setup and it works great for calling a single phone by setting
incomingcalllimit=1 and I can do something like:
exten 100,1,DIAL(SIP/100,,20)
exten 100,102,DIAL(SIP/200,,20)
exten 100,203,DIAL(SIP300,,20)
Does anyone have this setup? Or is it possible for a multiple phones to
register to the same extension (i.e. office mail number)?
Thanks.
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[Asterisk-Users] Connecting Asterisk to GSM
Hi List, I was wondering if there was any device I could use to connect * to GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, cheap is better :-) Any tips on this? Cheers, Jean-Michel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk on FC3
snip I have installed Asterisk on FC3 with all the current patches on an AMD Opteron 64 bit platform - this weekend I will transfer all of my working configs and cards from my old Asterisk box to the new box - I will report any issues that may arise. BTW - the AMD is one fast box - compiling Asterisk and all associated components from cvs/scratch took about 2 minutes. /snip I'm interested to hear how asterisk performance is on the Opteron. We've had great success with Opterons in server environments. -Ron___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI Card not recognized
Dear all i am using Fedora Core 2 . i have Planet BRI TA with HFC chipset ( hisax ) i can easyly connect to internet using BRI but this card is still not recognized by asterisk i am using i4l driver . some people suggest i should try bristuff from junghanns.net any ideas ? Thanks and Regards Talha ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Asterisk on SuSE 9.1?
Hi, On Tue, 14 Dec 2004 at 09:33, Rick Green wrote: (WHy is this?! I've noticed it with every sound card I've ever tried, and it infuriates me that I have to deal with feedback from the analog loopback in the sound card! Supposedly these soundards are full-duplex, so why are they looped by default, instead of keeping the inputs and outputs totally separated!) maybe because the sound cards are also being used for karaoke. ANybody know how to do an alsa.conf or set a mixer to fix this? Turn down the controller for the mic completely. It only controls how much from the mic's signal goes through the analog mixer directly to the speaker or line output, but has usually nothing to do with capturing. Then make sure that you have selected the mic as the input for capturing, and use the input gain controller to adjust the recording level. Depending on the sound card the mixer might have additional controls like a switch for a 2nd microphone or a mic boost switch that might need to be adjusted to get capturing from the mic working properly. cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI Card not recognized
Hi, Il giorno gio, 16-12-2004 alle 21:59 +0400, Muhammad Talha ha scritto: Dear all i am using Fedora Core 2 . i have Planet BRI TA with HFC chipset ( hisax ) i can easyly connect to internet using BRI but this card is still not recognized by asterisk i am using i4l driver . don't use i4l. is only a latency generator (ie you'll experience bad echo issues) some people suggest i should try bristuff from junghanns.net yes, go with that. We've bristuff running smoothly here with hfc based cards. Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calculating required bandwidth
I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, /edg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisk to GSM
Hi, Il giorno gio, 16-12-2004 alle 21:47 +, Jean-Michel Hiver ha scritto: I was wondering if there was any device I could use to connect * to GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, cheap is better :-) sure, mainly you can use gsm boxes with pstn to gsm interfaces. for example: for 1 gsm chan, you can use a box with an fxs interface on it, and can be connected to * via a single x100p (one fxo interface) Or for multi channels, you can go with a bri-gsm box, and interface it to * via a bri card (junghanns.net drivers) or even pri, with 16 or more channels (connected to the * with a pri card, ie te110p) or even sip... no card on the * box, but connected via a sip voip link. www.2n.cz has some of these products, but there're tons of them out there. prices? dunno exactly, the only that I'm aware of is that a bri - gsm (2 gsm chans) is something like 800 ¤ Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom FX Video Unit - asterisk-oh323
I'm installing an office in a couple of weeks that will have some nice Polycom FX video units in the conference rooms. I'm thinking that with asterisk-oh323 http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2 I should hopefully get the ability for phone users to dial an extension and participate in video conferences, or just simply phone conference with users in the room (would be able to use the multiple high quality mics that the Polycom has, and avoid purchasing a separate conference phone). Any tips / suggestions? I'm unfamiliar with the asterisk-oh323 stuff. Regards and TIA, -Ron___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sox-12.17.6
does this version work? after the asterisk MIXING of files i have a file of dead air. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone connected to 7960 with console cable for setup?
On Dec 16, 2004, at 10:34 AM, Randy MacKay wrote: I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. I don't think the 7960 has a console port on it, just ports for Ethernet (2), handset, headset, and 7914. The 7914 port is probably a serial link of some sort, but you won't get where you want by trying to talk to it via hyperterminal. I'd suggest trying a different DHCP server or adjusting its settings. It worked just fine for me out of the box. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calculating required bandwidth
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps equals 24 channels. This is also known as a PRI. -Matthew - Original Message - From: Ed Greenberg [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 16, 2004 11:45 AM Subject: [Asterisk-Users] Calculating required bandwidth I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, /edg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calculating required bandwidth
On December 16, 2004 01:52 pm, Race Vanderdecken wrote: The quick tyrannical answer, And wrong -- I am taking the time to correct it not so much to slam you but more for list posterity -- just because the codec rate is 64kbps doesn't mean that's what's actually on the wire, even if you ignore signalling. Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 each T1 has 24 channels of 8 bit data plus one frame bit. 24*8+1 = 193 bits per T1 frame. Frames are sent 8000 per second. 8000*193 = 1544000 bits per second. There's your T1 raw rate. You can't use that frame bit for yourself so 24*8*8000 = 1536000 bits per second. That's your T1 data rate; that's what you can actually use. Now. Running IP on a T1 you have certain overheads. UDP frame overhead is 4 bytes, plus your TCP overhead of 12 bytes, for a total of 16 bytes (128 bits). G.711 is 64kbps data rate, but Asterisk sends only 20ms per packet in an attempt to balance data throughput and effect of lost packets. so 64kbps / 50 is 1280 bits of audio per packet, plus 128 bits of overhead for 1408 bits per packet. 50 of these per second of audio gives you 70400bps for one second of G.711 VOIP audio. so now take your T1 data rate of 1536000bps and divide your audio rate into it for an answer of 21 channels of G.711 VOIP audio. Now that was straight UDP audio -- there was no signalling overhead and it wasn't SIP RTP. RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) Regards, Andrew the tyrant's tyrant Kohlsmith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup?
Randy, Is it a new unit? The only reason I ask is that hitting the settings button should let you straight in. There is an Rs232 port on the bottom - however not oversure what it's used for on the 7960's. The reason I as wether it's new or not is that it might need firmware resetting as per the cisco information (not immediately to hand). If you can see the menu's and just chance change the setting, I think it's something like *# or **# to allow change. Sorry if that's suck egg territory - just trying to cover anything obvious which is easily missed!! Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: 16 December 2004 18:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/ciscoconsole9.html DB9F RJ45 Receive Data2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground 5 4 Ground 5 5 Data Set Ready 6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 ==Pin 6 Pin 3 ==Pin 4 Pin 4 ==Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9FRJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] working with big blocks of msn's
On Thu, 2004-12-16 at 19:04, Sebastian Buntin wrote:
Hi!
I have to Set up an asterisk Server with a Diva Server PRI E1-30M.
Capi, asterisk, etc. everythink works.
my problem is the handling of the MSN's.
say, we have the block (without area-code..) 4321-0 to 4321-4999
between this numbers (including em) every MSN is possible.
do I have to add all MSN's i need (several hundrets) to the capi.conf?
No, you add the following lines instead:
incomingmsg=*
msn=4321
isdnmode=ptp
incomingmsn or msn in capi.conf would not work with more than 5 or 6
numbers. Anyhow this is actually in the common sample of capi.conf.
Please check that again.
then the routing to SIP-Phones shall be based on the MSN-Configuration.
means, if someone dials 4321-1000 the call shall go to SIP/boss
and 4321-1001 to SIP/secretary
and so on.
is this just by adding an
exten = 1000,1,Dial(SIP/boss)
to the context set in the /etc/asterisk/capi.conf?
context in capi.conf only points to a context in extensions.conf
The exten = line goes into the context in extensions.conf
and what to do, so that, if the boss calls out the MSN of the secretary
is shown?
and if the secretary calls out also their MSN is shown?
exten _XXX.,1,Dial(CAPI/:${EXTEN})
Where is the MSN you want to dial (including prefix here).
Slán leat,
Martin List-Petersen
Dublin, Eire
(contact info on -- http://www.marlow.dk/)
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Re: [Asterisk-Users] Connecting Asterisk to GSM
On Thu, 2004-12-16 at 21:47, Jean-Michel Hiver wrote: Hi List, I was wondering if there was any device I could use to connect * to GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, cheap is better :-) What you are looking for is something like the Ateus GSM to PSTN or ISDN gateways (http://www.mobilecomms-technology.com/contractors/gsm/2n_tele/) Cheaper would be some gsm to pstn adapter, that you can connect to the cellphone. Check the archives of the asterisk-users for that, because it's something, that commonly has been asked before. Another alternative would be chan_bluetooth, (http://www.crazygreek.co.uk/content/chan_bluetooth), but that is in a state far from working. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: iax.cc hosts - want to do some traceroutesbefore buying
For me it's iax2.sixtel.net... So far, so good. -Original Message- From: Kristian Kielhofner [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: iax.cc hosts - want to do some traceroutesbefore buying Sorry about this, but do any users have more detailed iax.cc information? Will they do trunking? What are the hosts that I will be logging into? I want to make sure that they will work well for me, and I would like to do some traceroutes to make sure that they are close! Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calculating required bandwidth
On December 16, 2004 02:00 pm, Damon Estep wrote: A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no compression (g.711) consists of 64k plus IP protocol overhead for a total bandwidth or 80 to 90k required per uncompressed channel. So a IP T1 carrying VoIP without compression has lower capacity that a Voice T1. A t1 for voice typically carries 23 b channels and 1 d channel, so 23 conversations not 24. Voice channelized T1 (also known as CAS T1 in Canada) is 24 channels. PRI is (simplified explanation) out of band signalling on a DS1, but uses 1 channel for signalling (it's out of band now, so it has to go somewhere) so you get 23 channels of voice and 1 for the signalling. Data T1 carrying VOIP traffic will be able to handle about 21 channels of G.711 RTP audio due to RTP and IP overhead, and does not include SIP/H.323 signalling, although the signalling overhead should be able to fit in the remainder of the T1. Uncompressed the answer is probably closer to 15 to 18 RTP streams across a dedicate T1 IP link. That few? I would be surprised if the signalling overhead is that enormous. With 21 channels of RTP G.711 audio I have 64kbps of bandwidth available, at least according to my calculations. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[SPAM] Re: [Asterisk-Users] working with big blocks of msn's
On 16.12.2004 Martin List-Petersen Wrote [EMAIL PROTECTED]: then the routing to SIP-Phones shall be based on the MSN-Configuration. means, if someone dials 4321-1000 the call shall go to SIP/boss and 4321-1001 to SIP/secretary and so on. is this just by adding an exten = 1000,1,Dial(SIP/boss) to the context set in the /etc/asterisk/capi.conf? context in capi.conf only points to a context in extensions.conf The exten = line goes into the context in extensions.conf ah yes, stupid me. I meant extensions.conf. so, what to put in the extens = ? like I wrote above? or do I have to write the full MSN here? exten = 1000,1,Dial(SIP/boss) or exten = 43211000,1,Dial(SIP/boss) or something completely different? thanks again! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calculating required bandwidth
Andrew Kohlsmith wrote: RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) And you can't run straight IP over a T1 circuit either; it's usually framed in HDLC frames. There's a little more overhead for you G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Boss wants background music!!!!
Satchid wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. How about modifying the chan_agent stuff? Right now, if an agent logs into a queue, he hears music until a call comes into him. So you have the option of making a queue for every phone (which wouldn't be all that great) or creating a new application that copies some of the agentlogin functionality, perhaps hanging up when a new call comes in? The bit stream is an MP3 file of 8 Kbs. At the server it might be at the maximum 570Kbs if it has to send it individually to each telephone. The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone. But the music isn't sent as an MP3 to the phone, it's sent using whatever codec you're using. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no compression (g.711) consists of 64k plus IP protocol overhead for a total bandwidth or 80 to 90k required per uncompressed channel. So a IP T1 carrying VoIP without compression has lower capacity that a Voice T1. A t1 for voice typically carries 23 b channels and 1 d channel, so 23 conversations not 24. If you use compression on the VoIP traffic you gain capacity, but loose CPU performance as the RTP data stream has to be transcoded by *. If compression is used, and the box has the CPU power, significantly more than 23 is the answer, probably limited more by then number that your * can setup, transcode, and tear down. The exact answer depends on your use and can only be determined through testing. Uncompressed the answer is probably closer to 15 to 18 RTP streams across a dedicate T1 IP link. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Thursday, December 16, 2004 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calculating required bandwidth I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, /edg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] native MOH with Asterisk 1.0.3
Does anyone have a WORKING native MOH patch for Asterisk 1.0.3? Thanks! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup?
I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy You have to enable the console with the phone programming. 2 things unlock the settings by pressing **# settings then you can go into network setup and keep scrolling down until you see alternate tftp yes/no. Make sure that it says yes. Save and reboot phone then you should be able to set the tftp server by pressing **# settings then scroll down to the TFTP server and enter in the correct ip address, save and reboot. If this is a newer SIP version you can press the settings button press the digit 9 then enter the password cisco Henry Devito Telephone Connection, Inc Network Design / Implementation Phone: 402.330.7510 Fax:402.330.8586 Toshiba CTX/DK/Stratagy Certified Cisco Certified Internetwork Expert (CCIE) Voice ( VoIP) Cisco Certified Internetwork Expert (CCIE) Routing and Switching MCSE Microsoft Certified Systems Engineer RHCE Red Hat Certified Engineer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calculating required bandwidth
You can encapsulate it as ppp, still some overhead, but less I think than HDLC. Ed - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 16, 2004 1:36 PM Subject: Re: [Asterisk-Users] Calculating required bandwidth Andrew Kohlsmith wrote: RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) And you can't run straight IP over a T1 circuit either; it's usually framed in HDLC frames. There's a little more overhead for you G Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxs causing constant CPU spikes
Can anyone confirm whether other Digium cards/drivers especially the Wildcard TE410P have sililar problems? On Thu, 16 Dec 2004 07:06:07 -0700, Michael Welter [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On December 15, 2004 09:27 pm, Michael Welter wrote: Yes? Is there a workaround, or do I tell my customer to go find something else? Are the CPU spikes causing you trouble or did you just notice them when poking around? I see CPU spikes but don't have any issues at all (noise or otherwise). -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Not just trouble but a disaster. I'm trying to build a spandsp (fax) application for a customer, and, whenever a process holds an interrupt too long, the fax transmission aborts and garbage is printed by a fax machine. -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxs causing constant CPU spikes
On December 16, 2004 03:29 pm, Eric Bishop wrote: Can anyone confirm whether other Digium cards/drivers especially the Wildcard TE410P have sililar problems? I do not have these spikes on a TE405P running ISDN PRI one span 1 and CAS T1 on the other three. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] native MOH with Asterisk 1.0.3
Kevin P. Fleming wrote: Kristian Kielhofner wrote: Does anyone have a WORKING native MOH patch for Asterisk 1.0.3? We are running the last version I posted to Mantis, coupled with twisted's moh stop patch that's also in Mantis, and it seems to be working fine. However, the patch I posted was not made against the 1.0 branch, so it's possible it won't work there; I don't think there would be a problem using it with 1.0.3, though. Kevin, Thanks for the quick reply. I was actually following your and anthm's patch, but it will not apply to 1.0 code... What is the bug ID for moh stop? -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup?
When I push the settings button, nothing happens. I never get a chance to put in the password. I think the previous owner may have messed up a firmware upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Cook Sent: Thursday, December 16, 2004 11:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup? Why can't you use the settings button? If you know the password (or using the default password) you should be able to unlock the phone and do a hard reset... Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] om] On Behalf Of Randy MacKay Sent: Thursday, December 16, 2004 1:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/cisc oconsole9.html DB9F RJ45 Receive Data 2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground5 4 Ground5 5 Data Set Ready6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 == Pin 6 Pin 3 == Pin 4 Pin 4 == Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9F RJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 _ __ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aste risk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste risk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
Interesting requirement. Depending on your site, you may consider alternative solutions. If you have individual offices, I suppose the PBX route would be the best way to go; however, if you have a shared space (cube-farm, call-room, whatever), maybe you can share the music source? In that case, I'd look into slimp3s with a slimserver, and an inexpensive shelf-system as the amplifier. www.slimdevices.com would have some more pointers. If you have dropped ceilings (as must businesses do), you could even run ceiling speakers and hook them up to a slimp3 or other music source via an inexpensive car-amplifier. Oh yeah, multiple slimp3s can be synchronized. -Original Message- From: Satchid [mailto:[EMAIL PROTECTED] Sent: Thursday, December 16, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] My Boss wants background music Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server it might be at the maximum 570Kbs if it has to send it individually to each telephone. The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone. Please, is there a way to get this done, otherwise I have to say goodbye to Asterisk (unless my boss gives in). Thank you all Willy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
Thank you Peasants, In general the original question was answered. I am software guy, if the network slobs can't fit all the data in the pipe that is not my problem. The basic idea in the answer was that you can get more calls by using compression; much like the automobile manufacture's gas mileage may vary. Also remember that a telephone conversation is 2/3's silence. ( I speak, silence, then you speak. See the book at bought on Amazon 4 years ago but can't remember the name of the book.)IP only sends the data when there is noise versus the T1 which is a constant TDM stream. So I predict in testing with good VoIP equipment you can get more then 24 G.711 calls per T1. So take that and comment. You should be able to get more VoIP calls, my prediction is 40 G.711 well behaved calls with silence suppression per T1. Why else would the Baby Bells move to VoIP? But it is nice to know there are some intelligent folks monitoring the list, thank you. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 16 December 2004 14:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calculating required bandwidth On December 16, 2004 01:52 pm, Race Vanderdecken wrote: The quick tyrannical answer, And wrong -- I am taking the time to correct it not so much to slam you but more for list posterity -- just because the codec rate is 64kbps doesn't mean that's what's actually on the wire, even if you ignore signalling. Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 each T1 has 24 channels of 8 bit data plus one frame bit. 24*8+1 = 193 bits per T1 frame. Frames are sent 8000 per second. 8000*193 = 1544000 bits per second. There's your T1 raw rate. You can't use that frame bit for yourself so 24*8*8000 = 1536000 bits per second. That's your T1 data rate; that's what you can actually use. Now. Running IP on a T1 you have certain overheads. UDP frame overhead is 4 bytes, plus your TCP overhead of 12 bytes, for a total of 16 bytes (128 bits). G.711 is 64kbps data rate, but Asterisk sends only 20ms per packet in an attempt to balance data throughput and effect of lost packets. so 64kbps / 50 is 1280 bits of audio per packet, plus 128 bits of overhead for 1408 bits per packet. 50 of these per second of audio gives you 70400bps for one second of G.711 VOIP audio. so now take your T1 data rate of 1536000bps and divide your audio rate into it for an answer of 21 channels of G.711 VOIP audio. Now that was straight UDP audio -- there was no signalling overhead and it wasn't SIP RTP. RTP has 12 octets all its own, and still need 12 bytes of IP overhead, so it is actually costlier: I'll spare you all the calculations but it's 20 channels of SIP G.711 audio per T1, likely with enough room for signalling. :-) Regards, Andrew the tyrant's tyrant Kohlsmith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Termination
On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote: On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote: One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the phone, and one that works well with Asterisk. You've gotta be kidding, VOIP providers are trying to regulate who you can call? Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over SIP, IMO it's just better. Thanx, I will look into these providers. This is an exerpt from Packet8's Terms of Use statement. I've edited it for space, but I've tried to retain the context: -- PERSONAL USE. 8x8's Service Plans for residential subscribers that offer unlimited minutes of PSTN calls (Unlimited PSTN Plans) are for the reasonable personal residential use of End User only. End Users of Unlimited PSTN Plans shall not use the Services for commercial or governmental purposes or for profit or non-profit activities, including, but not limited to, home office, business, sales, tele-commuting, autodialing, continuous or extensive call forwarding, continuous connectivity, fax broadcast, fax blasting, telemarketing or any other activity that would be inconsistent with personal and residential usage. 8x8 reserves the right to immediately terminate or modify the Services of any End User using Unlimited PSTN Plans if 8x8 determines, in its sole discretion, that End User is not using the Unlimited PSTN Plans for End User's reasonable personal residential use. -- Now I agree with their policy on fax-blasting, etc. But according to them, I can't use my own phone for charity work? I work at a national lab; would my wife be alowed to call me at work? Or would the be a governmental purpose? It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm conducting a business with my phone, they can terminate my service, or increase the price of it. I'm trying to make an issue out of this because I think it needs to change and I'm hoping people who are affiliated with these providers are reading this. I was going to go with Packet8. I was going through the final checklist before subscribing when I came accross this fascist policy. Sure, I can go with a business plan, but that would cost me $39.95. That's $5 more than I'm spending for an analog phone line! Part of the reason for me to go with VoIP is to become Quest Free. But suddenly, Quest is starting to resemble the Boy Scouts when compared to the types of usage policies I'm seeing from some of the VoIP providers. Sorry for the rant, but I hope you understand. -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] native MOH with Asterisk 1.0.3
That moh_stop is for the MusicOnHold application ONLY. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, December 16, 2004 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] native MOH with Asterisk 1.0.3 Kevin P. Fleming wrote: Kristian Kielhofner wrote: Does anyone have a WORKING native MOH patch for Asterisk 1.0.3? We are running the last version I posted to Mantis, coupled with twisted's moh stop patch that's also in Mantis, and it seems to be working fine. However, the patch I posted was not made against the 1.0 branch, so it's possible it won't work there; I don't think there would be a problem using it with 1.0.3, though. Kevin, Thanks for the quick reply. I was actually following your and anthm's patch, but it will not apply to 1.0 code... What is the bug ID for moh stop? -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Steps to configure D/41EPCI card
Hi, Somebody can give me the necessary steps for configuring a D/41EPCI in Asterisk. Thanks in advance, Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamically Choose Codec for Bandwidth Management
Is there any way to set Asterisk to choose what codec to allow for a new call based on current usage? In other words... be able to define a max number of ulaw calls, then after that only allowing g729? The idea here is that in general, a T-1 should be enough for our offices to have phone + citrix + some video (got good QoS in place already). But for usage spikes, user experience would be kept good if we could shift it into using g729. -Ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
