Re:[Asterisk-Users] Problem with X101P
Scott Stingel wrote: Some questions: What country are you in? Is there anything else connected to the line from the PSTN? It sounds like you have a marginal condition, such as insufficient loop current perhaps. Do have any features, such as call waiting, on the line? Do you know how far you are from the central office? Do you have another line you can switch to and try the same card? Does the Red alarm occur at the moment the call is disconnected, or afterward? Sorry for late reply. Answers: --I am in Bangladesh. --No there is nothing else connected with my PSTN line. But, in future that line would be connected with the Fax simultaneously. -- No call waiting feature on the line. But in zapata configuration this feature is true. -- Sorry I didn't get that question. --Yeah I have other two lines and I have checked with those lines with the same card. Same thing happens for those. -- The Red Alarm occurs when I am connected and having conversation with other party. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with ser to share the load
Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with ser to share the load
Hello Deepak, 1. don't post multiple times. it's annoying. enough said. 2. run asterisk in verbose mode (start it with asterisk -vgc), place a call from a SIP endpoint behind SER to the asterisk server, and see what happens in the asterisk CLI. 3. if you don't see anything there, get ngrep and place a call from the SIP endpoint while running ngrep SIP and post the output. 4. are asterisk and SER on the same machine? 5. if all else fails put autocreatepeer=yes in your sip.conf - this has bad security consequences, but it is useful for debugging. -yair On 12/2/04, Deepak Dhiman [EMAIL PROTECTED] wrote: Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Paul Hales wrote: It now works - but only in the latest (1.5+) firmware releases. Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dutch SIP or IAX numbers
Hi, -Original Message- How knows where I can get a Dutchphone number for asterisk? Pilmo is not delivering one for home use. I think you are physically outside the netherlands, right ? Would you care for an 087 number ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA2000 dialplan vs Asterisk dialplan
Steve Prior wrote: I've got a Sipura SPA2000 ATA basically working (I can place calls between the extensions plugged into each of its ports) and part of that was setting up the dial plan on the SPA2000 to match the one in Asterisk. This seems like a pain to deal with long term and I don't know what exactly the dial plan built into the SPA2000 does for me over the one in Asterisk. Is there a way to disable the use of the SPA2000 dialplan so I don't have to keep it in synch? Or is there some reason why it would be a bad idea for me to do so? Sure just put x. as your dial plan and any number will be accepted. The catch is you'll have to wait for the Short (Long?) Digit Timeout to pass before the call goes to asterisk for processing. If the SPA has an idea of what digit combinations are accepted it will wait until it has a match and send the call along at just the right time. No delays waiting the digit timer to expire. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to disconnect a call manually
soft hangup channel name -Abhishek Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/2/05, Asterisk guy [EMAIL PROTECTED] wrote: 1 after giving command oh323 show channels, i want to disconnect a call, is there any command to disconnect a call? 2 how asterisk kill a hung/dead call ? for most commercial softswitch, there are a setting for maximum duration for a call. they will hang up it l if its duration reachs the limit. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to use addpac-ap200 (sip | h323)
Hi, I am testing an AP200 from addpac i m trying to make it register with Asterisk. It manages 3 protocols (sip, h323 and mgcp). If i use sip, i keep getting this messages: May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.202' May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.202' the config in the sip.conf is this: [108] type=friend host=dynamic username=addpac1 secret=test reinvite=no qualify=1000 canreinvite=no dtmfmode=rfc2833 context=toll-access callerid=addpac 108 allow=all disallow=g723.1 [83] type=friend host=dynamic username=addpac1 secret=test reinvite=no qualify=1000 canreinvite=no dtmfmode=rfc2833 context=toll-access callerid=addpac 108 allow=all disallow=g723.1 The addpac config for the sip is this: ! VoIP configuration. ! ! Voice service voip configuration. ! voice service voip busyout monitor gatekeeper busyout monitor voip-interface ! ! ! Voice port configuration. voice-port 0/0 caller-id enable caller-id type bellcore ! voice-port 0/1 caller-id enable caller-id type bellcore ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 83 port 0/0 ! dial-peer voice 1 pots destination-pattern 108 port 0/1 ! ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target sip-server session protocol sip dtmf-relay rtp-2833 ! ! ! ! Gateway configuration. ! ! gateway ! ! SIP UA configuration ! sip-ua sip-username 108 sip-password test sip-server 192.168.1.197 register e164 Thanks to vpp in [EMAIL PROTECTED], that said to commend the bind line in sip.conf, the extension 108 could register, but not the 83. And finnally i could make ring the extension, but after a minute or so, some kind of timeout happends and then i start getting the same failed lines again and i cant connect again. -- Using h323, from asterisk/channels/h323 of the cvs downloaded tree (i also downloaded the right versions of pwlib and openh323), it compiles with no problem (takes too much time to compile openh323, as usual). Using the default h323.conf.sample that comes in that directory, i found something strange. Again, i couldnt register configuring my ap200 with the h323 protocol. And doing a netstat i got this: [EMAIL PROTECTED] asterisk]# netstat -lpn | grep asterisk tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 31178/asterisk tcp0 0 0.0.0.0:17200.0.0.0:* LISTEN 31178/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 31178/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 31178/asterisk udp0 0 192.168.1.197:2427 0.0.0.0:* 31178/asterisk There are missing the other default udp ports for h323 (1721, 1719, 1718). I tryed using netmeeting as well, but no susses. A tcpdump shows this: 192.168.1.200.1660 192.168.1.197.h323gatestat: udp 73 192.168.1.197 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] 192.168.1.200.1660 192.168.1.197.h323gatestat: udp 73 192.168.1.197 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] 192.168.1.197.ntp 192.168.1.200.ntp: udp 48 (DF) [tos 0x10] 192.168.1.200.ntp 192.168.1.197.ntp: udp 48 192.168.1.200.1660 192.168.1.197.h323gatestat: udp 73 192.168.1.197 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] The addpac is not the problem in this case, as it can register in a gatekeepr as netmeeting does too. --- I am about to use MGCP, but need more documentation about it, i've never used it before, so i dont have a clue about it, but i'll try it too. Any hint would be apreciated Regards, -=Raul=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323
Hi I've installed successfully: - PWlib v1.6.7 library -Openh323 v1.13.5 library -asterisk-oh323 v0.6.5 and so the modules chan_oh323 is installed successfully Now I try to install chan_h323 First question: is this necessary? I edit the Makefile in the directory /usr/src/asterisk-1.0.7/channels/h323 to point to the right includes directories I do makeand I've the following error: make:***[ast_h323.o] Error 1 Have you some suggestions? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback() stops working.
On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. I have heard from 5 or so people about this problem. I run CVS STABLE (almost the same as 1.0.7) and had none of these issues. I wish I could help you, but I wanted to let you know that this is not a general problem. Well a quick reinstall of Asterisk solved the problem, but I hope it doesn't happen again :) Thanks for the heads up Eric. ~sm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme and a timing source
All, I seem to be confused :( Meetme won't work with the message That is not a valid conference number, please try again even with the simplest of configurations. Having trawled the list archives, wiki and harrased people on #asterisk I've come to a dead-end. I compiled ztdummy last night from source and it inserts without any errors. I also have a Digium TE110P card in the server although it currently isn't plugged into anything. Am I able to use this zaptel card for timing even though it isn't 'live'? I haven't configured a lot in /etc/zaptel.conf or /etc/asterisk/zapata.conf uname -r: 2.4.27-2-386 lon0asterisk01:~# lsmod Module Size Used byNot tainted ztdummy 1688 0 (unused) wcte11xp 20352 0 (unused) zaptel216608 2 [ztdummy wcte11xp] hisax 473648 0 (unused) isa-pnp25552 0 [hisax] isdn 112204 0 [hisax] slhc4144 0 [isdn] ehci-hcd 14764 0 (unused) usb-uhci 19504 0 [ztdummy] usbcore52268 1 [ehci-hcd usb-uhci] ide-scsi8272 0 piix7784 1 e1000 57676 1 ide-disk 12448 0 ide-detect 288 0 (unused) ide-cd 27072 0 cdrom 26212 0 [ide-cd] ide-core 91832 0 [ide-scsi piix ide-disk ide-detect ide-cd] rtc 5768 0 (autoclean) ext3 65388 3 (autoclean) jbd34628 3 (autoclean) [ext3] sd_mod 10764 8 (autoclean) megaraid2 28616 4 (autoclean) scsi_mod 86052 3 (autoclean) [ide-scsi sd_mod megaraid2] unix 12752 80 (autoclean) Any pointers greatfully received. ~sm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can anyone recommend some hardware for UK use?
Same as subject!! I want to build my own * box - and i would like some recommendations for the FXO cards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: 2) There isn't anything like what you want. I know, I want the same thing. There is no phone out there that will do this with any protocol that asterisk uses. This is the one major failing of asterisk ( and voip in general. I smell an oportunity for a phone manufacture ), and what keeps it out of a lot of places. It's alright, you can come out from under your rock now The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diffrence bewteen FXO and FXS
Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic phone groups.
On Sat, 2005-04-30 at 15:25 +0200, Joris Vandalon wrote: Hi, I am looking for a way to dynamicly put phones in a group so if someone calls an extentions everyone's phone who's member of the group will ring. Queues are not an options because as soon a call comes in to a queue there is no getting out. I want to let the phones ring and after a period of time stop trying and continue to voicemail for example. Can someone provide me with some hints or examples getting this done? Why not use the timeout parameter to queue? Along with the joinonempty=no and leavewhenempty=yes etc that have all been discussed within the last week show application queue and also read the queues.conf, and refer to the wiki for even more details. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
On the subject of phones.. The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. How about phones that can indicate if an extension is busy or not - eg, Busy Lamp Field - can anyone point to a list of phones that implement this? (The 'Asterisk standard extensions' page on voip-info mentions the Snom, Polycom and Sayson(???) phones. Any *cheaper* ones? --Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller Hears Ring During Attended Transfer?
On Sun, 2005-05-01 at 22:43 -0500, Eric Wieling aka ManxPower wrote: Eric Wieling aka ManxPower wrote: Daryll Strauss wrote: I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. A call comes in to my asterisk box via SIP (the Sipura isn't involved) and I answer it using an analog phone on the Sipura. I then decide to forward it to another phone. I flash the line, dial the new extension, and flash again. At this point I and the caller hear the destination phone ring. If I then hang up, the caller stops hearing the ring. Eventually the voicemail picks up and the caller hears the announcement and everything works as normal. It's just rather disconcerting for the caller to not hear anything. Is this an Asterisk bug, Sipura bug, or operator (me) error? Happens with Polycom phones as well. It may be related to this message when a transfer happens: Unable to handle indication 3 for 'SIP/0004f201e4b3-a-682d' I managed to 'resolve' this problem for a customer by simply continuing the music on hold instead of playing the ringing (by the m parameter to dial). Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling out through second server.
Hi, Im trying to set up the following situation. Ive got several inbound server en one server where we can make an outbound call. If a caller calling to a inbound asterisk server (say serverA) then I want to connect the call to the outbound server(say serverB) So an inbound caller will be connected to an outbound call. Ive read the www.voip-info.org about connecting two asterisk servers. But I couldnt succeed to get it working with an iax connection Maybe somebody got the same situation working on his asterisk server? Weve got static ip-addresses on al of our servers. Thanks in advanced Arjan Kroon email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323
[EMAIL PROTECTED] a écrit : Hi I've installed successfully: - PWlib v1.6.7 library -Openh323 v1.13.5 library -asterisk-oh323 v0.6.5 and so the modules chan_oh323 is installed successfully Now I try to install chan_h323 First question: is this necessary? No, it's or oh323 or h323. I suggest you to stay with oh323 -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Putting in an Application
Hi All! I am using Asterisk Stable 1.0.6 . Now I want to add another application like app_chanspy in it. I have downloaded its source file but how can I merge this application along with my already running asterisk ? Any comments suggestions are appreciated ... Thankyou, Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] config for call pstn from voip
hello,
newasterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone answer the call and i've echo please may somebody help me?
there is my config file
zaptel.conf
fxsls=4#X100Pdefaultzone=frloadzone=fr
zapata.conf
[channels]language=fr relaxdtmf=yesimmediate=nocontext=pstnsignalling=fxs_ls;X100P;Cidsignalling=v23;Cidstart=polarity;usecallerid=yes;callerid="fone" 60
extensions.conf
[general]static=yeswriteprotect=no
[pstn]exten = 19100,1,dial(SIP/799SIP/788)
exten = 788,1,dial(SIP/788:5060)
exten = 799,1,dial(SIP/799:5060)
exten = _00N,1,dial(Zap/4/${EXTEN:1}); i want to call analog phone
exten = _6059,1,dial(SIP/799)exten = s,1,dial(SIP/799SIP/788);here i can recieve analog calls
regards.
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RE: [Asterisk-Users] Diffrence bewteen FXO and FXS
Hi, First, do forgive any syntax or language errors as English is not my mother tongue. To make a 'long' story short, - let's look at a phone home. The public switch (the very big machine at your telco) provides power to the line. This power gives energy to your phone. This is how a 'standard' phone can work without external power adaptor. So, the public switch sends power, the private phone receives power - lets look at a small office there is a pbx (private box) it is connected to the public phone line(s) it receives power on those lines from the public switch (also it will use the 110-240V power lines for real power) it is connected to several insides phones. It provides power to those inside phones like the public switch does. So, some interfaces receive power, others send power. You can also think as male and female plug. You plug the male with a female Likewise, you connect a fxo to fxs. 'Fxo to fxo' or 'fxs to fxs' will not work and in one case even bring problem as the two are sending power. The public switch is most of the time rather protected against most things, but your card not always. So you can send some circuit to silicon heaven. Some googling will provide similar (or better) explanations Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Mr AG!! [mailto:[EMAIL PROTECTED] Sent: lundi 2 mai 2005 11:12 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Diffrence bewteen FXO and FXS Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Power Connector
I have a TDM400P I am trying to install but I need a power connector extender to be able to get power into the card. In the meantime can the card run without the power connector if it has only one module on it? The power connector is only needed if you have fxs modules installed. The fxo does not use that connector/power. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Problems with TDM400P card
On Mon, May 2, 2005 1:12 am, Geoffrey Sachs said: Thanks for the info. What hard drives are you using ide or serial ata. Does it make a difference. Thanks There have been some references recently regarding disk drive types relating to tdm400 noise problems. Has anyone established there is a correlation between drive hardware and noise? If this is the case, it may be indicative of marginal interrupt timng performance on that hardware. FWIW the system described earlier has a single sata drive attached to the Intel 925XCV mb on-board controller, from dmesg: atapci1: Intel ICH6 SATA150 controller The drive is a Seagate ST3250823: ad4: 238475MB ST3250823AS/3.01 -kim -- [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Diffrence bewteen FXO and FXS
Mr AG!! wrote: Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? FXO card = plug to phone line FXS card = plug to regular phone FXS provides power to phone. FXO takes power from phone line. Don't plug PSTN in your FXS card or you could fry it... Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] increasing delay in meetme conference
Hello! I would like to use meetme app for audio conference, but I have a problem with the audio delay: it is increasing as time elapses. One way delay can reach 5 secs after 15 minutes usage. I have read all the mails in the list archives, and the open case for the bug: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003599 I have tried the code from the devel thread of CVS today, and the 'q' parameter, but the result was the same. There is nearly no delay when the participants join to the conference (2 SIP phones from the same LAN), but after some minutes the delay permanently increases. There is no problem when the phones call each other without meetme app. Asterisk has to transcode every time (gsm-alaw), but the thranscoding time between the two codecs is 2-3 milisecs, and the cpu usage is less the 1%. It is also true in case of using meetme. Has somebody a good solution for this problem? Thanks, Miklos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
Any *cheaper* ones? --Rob You want a _cheap_ reception phone? I don't think you are going to get this. Chris Mason www.anguillaguide.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loading ztdummy Stops MoH but Conference Works - VmWare
Hi Folks, I hope you can help !!! I have going arround in my [EMAIL PROTECTED] 0.9 to have the conference rooms and MoH working together. Everything works perfectly except this two features. I believe that all has to do with the ztdummy interface. After all installed I ran 'yum -y update'. Kernel was upgraded and therefore zaptel needed to be compiled. But I not only compiled zaptel but asterisk completely. Everytime I load ztdummy MoH and Playing files stops working, cannot listen any sound from *. On the other hand if I unload ztdummy I can hear MoH and all prompts but then I get "invalid Conference Number". It could be for the fact of vmWare not implementing USB, but it does. I can even mount my USB PEN. Can anybody help me? Thanks, Nuno ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
--- Tim Connolly [EMAIL PROTECTED] wrote: Is NAT=yes on, are you behind a firewall? Give us some connectivity details. Usually when you see maximum retries, its because you have one-way communications with the far end for some reason. Are you setting externip statically? THANKS, Tim! That nat=yes bit did the trick as far as the console messages go. Now I see a registration successful message on the console. (Is it normal for this to happen every few minutes? I haven't timed it but it looks like the registration is recurring every couple of minutes, maybe even once every minute.) Unfortunately, I still get only a busy tone when I dial the Canadian DID. When I dial the Canadian number, absolutely nothing happens on the console! Interestingly, even when I dial the local California number (I'm assuming it's still using IAX at that point), it seems to go into the unwelcome-calls extension (which I thought was sip-specific) and, according to the console, plays the congestion stuff which I never hear. Thanks again, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Diffrence bewteen FXO and FXS
Hi, FXS card = plug to regular phone FXO card = plug to phone line The trick I use is: FXO with a 'O' as in Office. This is where you plug your phone A FXO card emulates a phone (receives power) FXS with 'S' as in (public) Switch This is the part that gives power A FXS card emulates a switch (gives power) Plugging 2 FXS elements together is not a good idea Plugging 2 FXO elements together will not work but no problem otherwise Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dutch SIP or IAX numbers
Yes, I am physically living in Rio de Janeiro, but I am going back to Holland. I have at this moment a dutchphone connection with a 020 number. I think for people to call me that will be cheaper and easier to accept then a 087 number. How every people who would like to call me can not belief that they call me for local fee to Brazil. Greetings Han -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Monday, May 02, 2005 3:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dutch SIP or IAX numbers Hi, -Original Message- How knows where I can get a Dutchphone number for asterisk? Pilmo is not delivering one for home use. I think you are physically outside the netherlands, right ? Would you care for an 087 number ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions.conf dial plan
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a bigger set of
phones.
Basically, this is easy,
I can do this in the extensions.con with
[ISDN-in]
exten= 6201030,1,setcallerid(${CALLERID} ${CALLERID}|a)
exten= 6201030,2,dial,${UserGroup1}|20|t
exten= 6201030,3,dial,${UserGroup1UserGroup2}|60|t
exten= 6201030,4,Voicemail2(u6201030)
exten= 6201030,5,hangup
exten= 6201030,302,Voicemail2(b6201030)
But here is on major problem,
in step 2, after 20 seconds, the call on the phones in Group1 will be
terminated and then restarted in the bigger group (Group1Group2).
The problem with this is, during the transition is a time gap of a view
seconds on the phones from Group1. That means, if I lift up the head set
during this gape, then I can loos the calls on those phones.
Hence, I was wondering if I can set the dial proceadure such, that I
have the calls for 80 seconds on the phone Group1, and after 20 seconds
additionally on the phone Group2 without any interruption of the ringing
on the other phones.
Best regards
Georg P. Israel
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Re: RE: [Asterisk-Users] Problems with TDM400P card
Thanks for the info. What hard drives are you using ide or serial ata. Does it make a difference. Thanks There have been some references recently regarding disk drive types relating to tdm400 noise problems. Has anyone established there is a correlation between drive hardware and noise? If this is the case, it may be indicative of marginal interrupt timng performance on that hardware. FWIW the system described earlier has a single sata drive attached to the Intel 925XCV mb on-board controller, from dmesg: atapci1: Intel ICH6 SATA150 controller The drive is a Seagate ST3250823: ad4: 238475MB ST3250823AS/3.01 One of the easiest ways to determine whether any disk drive is impacting audio quality is evaluate the system in a no-load environment. (eg, process a single call with nothing else going on in the system including no swapping.) Then compare the audio to the same test repeated while generating large amounts of disk activity. (If I recall, 'hdparm -t' generates lots of disk activity.) Tests conducted this weekend (but incomplete right now) suggest the OS is doing something that impacts the TDM card specifically. Not sure what that is as yet, but likely to have something to do with the pci bus and/or interrupt handling. Seems the TDM card implementation (at least in the RHv9 distro) is not being serviced in reasonable timeframes. I modified the zttest.c app to display the length of time required to receive 8192 bytes of data from the card. In all cases tested thus far, the 8192 bytes are received in about 1.021000 seconds (21000 microseconds to late). That would suggest the data arriving from a TDM card will miss a frame of data roughly every ten frames. That has a serious impact on trying to run things like spandsp, but less of an impact on pure audio. The tests on this single system indicate that playing with the pci latency values had zero impact on the TDM timing. Also, suggestions involving 'udma2' on the drive had zero impact. That only confirms that if there isn't any disk activity, those parameters would have no audio impact. To help identify the source of the delays, I built a new system this weekend from scratch. When that is complete, I'll use it to compare the differences in motherboards, OS distro's, and maybe kernel versions. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
You want a _cheap_ reception phone? I don't think you are going to get this. Heh. I had a sneaking suspicion that was going to be the answer 8) --Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Diffrence bewteen FXO and FXS
Mr AG!! wrote: Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? For one thing, modem cards do not generate a ring voltage (they just pass it thru from the telco. Same for the dialtone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Adam Goryachev wrote: The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Regards, Adam I am currently having the same problem with our receptionist. We use 7960's, which I really like. The problem with it is that when you are trying to manage 6 lines with it, it has a tendancy to make you mess up. Example, you are talking on line 3 and about to transfer the call or put them hold when line 4 rings. The SIP image will move to line 4 and you inadvertantly answer line 4 instead of transfering line 3. It would be nice if it would stay on the current button and let you select the line you want as opposed to it just jumping around to whatever the newest call happens to be. The Skinny image was a little better in this respect. Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
Take a look at the Polycom 360 if you only nee 12 lines. otherwise look at the Snom 220 with a sidecar (up to a total of 3 side cars may be added for a total of 65 lines in the extreme need.) Max W . Blackmer, Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] txfax and Ghostscript 8.51
If the problem is with libtiff, its a problem with every version i've tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2) On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote: Me wrote: Hi all, I'm trying to use spandsp and asterisk to send faxes. To do so I am creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems to work fine, but when I create the tiff using Ghostscript 8.51 (or 7.06) txfax garbles the tiff and it comes through all messed up. First of all is this a known problem or is it just me. More importantly does anyone know of a way to fix this, I'd like to use 8.51 instead of 6.50. By the way, if it makes a differnece i'm currently running [EMAIL PROTECTED] but I've encountered the same problem with all the other asterisk builds i've tried It is really a change to Ghostscript or a related change to libtiff causing you problems. Libtiff is the usual suspect when FAX images go wrong. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite and callto:// syntax in webpages
Hi, does anyone know if x-lite supports the callto://name syntax on web pages as skype does? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CVS and bristuff-0.2.0-RC8a-CVS: no callerid
td wrote: -- Executing NoOp(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/tdhome) in new stack -- Called tdhome Same problem here. Any ideas? Deti ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Diffrence bewteen FXO and FXS
Here is an excellent document explaining the differences between FXO and FXS. http://www.google.com/url?sa=Ustart=5q=http://www.patton.com/technotes/fxs_fxo.pdfe=7385 Also you can look at Digium's site for their description, which describes it from a stand point of Asterisk as the PBX. http://www.digium.com/index.php?menu=fxsvfxo Why are there FXO cards, and FXS cards? What's the difference, and why is it needed? Modem cards, seem to be able to dial out, and receive calls, so why are these cards different? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: extensions.conf dial plan
In article [EMAIL PROTECTED],
Georg P. Israel [EMAIL PROTECTED] wrote:
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a bigger set of
phones.
Basically, this is easy,
I can do this in the extensions.con with
[ISDN-in]
exten= 6201030,1,setcallerid(${CALLERID} ${CALLERID}|a)
exten= 6201030,2,dial,${UserGroup1}|20|t
exten= 6201030,3,dial,${UserGroup1UserGroup2}|60|t
exten= 6201030,4,Voicemail2(u6201030)
exten= 6201030,5,hangup
exten= 6201030,302,Voicemail2(b6201030)
But here is on major problem,
in step 2, after 20 seconds, the call on the phones in Group1 will be
terminated and then restarted in the bigger group (Group1Group2).
The problem with this is, during the transition is a time gap of a view
seconds on the phones from Group1. That means, if I lift up the head set
during this gape, then I can loos the calls on those phones.
Hence, I was wondering if I can set the dial proceadure such, that I
have the calls for 80 seconds on the phone Group1, and after 20 seconds
additionally on the phone Group2 without any interruption of the ringing
on the other phones.
I don't have a proven answer, but here is an idea to try:
[ISDN-in]
exten= 6201030,1,SetCallerID(${CALLERID} ${CALLERID}|a)
exten= 6201030,2,Dial(${UserGroup1}Local/[EMAIL PROTECTED]|80|t)
exten= 6201030,3,Voicemail2(u6201030)
exten= 6201030,4,Hangup
exten= 6201030,302,Voicemail2(b6201030)
[ISDN-in-delayed]
exten= 6201030,1,Wait(20)
exten= 6201030,2,Dial(${UserGroup2}|60|t)
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: RE: [Asterisk-Users] Problems with TDM400P card
On Mon, May 2, 2005 8:24 am, Rich Adamson said: To help identify the source of the delays, I built a new system this weekend from scratch. When that is complete, I'll use it to compare the differences in motherboards, OS distro's, and maybe kernel versions. Very good Rich, the results of that work will be very interesting. Realtime scheduling modifications for Linux and FreeBSD are discussed on Mantis at: http://bugs.digium.com/bug_view_page.php?bug_id=0003203 Should you decide to evaluate Asterisk on FreeBSD, you might want to take a look at Staffan Ulfberg's excelllent contribution on the above site, with links to patches near the bottom of the page. The system described in my recent post was built from * CVS Head of 4-22-05 with these patches applied, with the exception of the changes which cause Asterisk to lower it's priority to normal after forking. The zaptel drivers for FreeBSD are from 4-26-05, downloaded from the Subversion repostory as described at: http://www.voip-info.org/wiki-FreeBSD+zaptel Patches to the zaptel drivers are described on the Mantis link above. -kim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:How to use ser with asterisk server for load sharing
Hi friends ! Can anybody help me that how to use ser with asterisk server so that ser can work like the front end of the asterisk and all other features of the asterisk can be used. I have tried the configuration given in asterisk-wiki/at+large but could not succeed, still my asterisk in not listening to ser or ser is not forwarding to asterisk. Thanks Deepak Dhiman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, April 28, 2005 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? The RJ11 plug fits perfectly into an RJ45 socket and only cares about the center-most conductors, which are the ones with the connection to the PSTN. Mojo Paul Shiflet wrote: I just received my TDM400 card from digium with 2 fxo and 2 fxs interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS phones. How do i interface my POTS phones with this; can i just crimp an RJ45 connection on the end of the phone cord? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] txfax and Ghostscript 8.51
Send an example TIFF file, and I will investigate. Regards, Steve Me wrote: If the problem is with libtiff, its a problem with every version i've tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2) On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote: Me wrote: Hi all, I'm trying to use spandsp and asterisk to send faxes. To do so I am creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems to work fine, but when I create the tiff using Ghostscript 8.51 (or 7.06) txfax garbles the tiff and it comes through all messed up. First of all is this a known problem or is it just me. More importantly does anyone know of a way to fix this, I'd like to use 8.51 instead of 6.50. By the way, if it makes a differnece i'm currently running [EMAIL PROTECTED] but I've encountered the same problem with all the other asterisk builds i've tried It is really a change to Ghostscript or a related change to libtiff causing you problems. Libtiff is the usual suspect when FAX images go wrong. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Problems with TDM400P card -correction to last post
On Mon, May 2, 2005 9:01 am, Kim Culhan said: Patches to the zaptel drivers are described on the Mantis link above. El wrongo kimster, they're described in this post to the asterisk-bsd list: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000719.html The patches are in this post: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000722.html -kc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE:How to use ser with asterisk server for load sharing
Fix the date on your PC. Deepak Dhiman wrote: Hi friends ! Can anybody help me that how to use ser with asterisk server so that ser can work like the front end of the asterisk and all other features of the asterisk can be used. I have tried the configuration given in asterisk-wiki/at+large but could not succeed, still my asterisk in not listening to ser or ser is not forwarding to asterisk. Thanks Deepak Dhiman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, April 28, 2005 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? The RJ11 plug fits perfectly into an RJ45 socket and only cares about the center-most conductors, which are the ones with the connection to the PSTN. Mojo Paul Shiflet wrote: I just received my TDM400 card from digium with 2 fxo and 2 fxs interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS phones. How do i interface my POTS phones with this; can i just crimp an RJ45 connection on the end of the phone cord? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phonejack PCI-card
Hi I am using Phonejack PCI card connected to analog phone. I've installed this card succesfully but i get no dial tone. Have you suggestions? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] Problems with TDM400P card
ide |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Geoffrey Sachs |Sent: Lunes, 02 de Mayo de 2005 12:13 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card | |Thanks for the info. | What hard drives are you using ide or serial ata. |Does it make a difference. | Thanks | Geoffrey Sachs |- Original Message - |From: Anton Krall [EMAIL PROTECTED] |To: 'Kim Culhan' [EMAIL PROTECTED]; 'Asterisk Users |Mailing List - Non-Commercial Discussion' |asterisk-users@lists.digium.com |Sent: Saturday, April 30, 2005 7:44 AM |Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card | | |Hows does this look? | |Opened pseudo zap interface, measuring accuracy... | |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8194 sample intervals 99.975586% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |8192 samples in 8193 sample intervals 99.987793% |--- Results after 13 passes --- |Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 | |Good enough and what do I need to check in order to make 100%? |What does the test actually measure? | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Kim ||Culhan ||Sent: Sábado, 30 de Abril de 2005 08:45 a.m. ||To: asterisk-users@lists.digium.com ||Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card || ||On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said: || I would also be interested in alternatives to the Tdm400p. I ||have had || endless problems with a tdm400p card not being able to get ||the zttest || numbers above || 99.975 and as a result not being able eliminate an ||intermitent but consistent echo. || I have tried to date 4 different motherboard and hardware ||combinations || as well as different linux versions to no avial.I would ||welcome some feedback on this. || ||Since there appear to be several combinations of hardware and |operating ||system which don't work well, here is a combination which appears to ||work fairly well: || ||Intel 925XCV mb || ||P-4 560 (3.6 gHz) || ||wcfxs0: Wildcard TDM400P REV E/F || ||FreeBSD 5.4-STABLE || ||zttest -v ||Opened pseudo zap interface, measuring accuracy... || ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ||8192 samples in 8192 sample intervals 100.00% ^C ||--- Results after 10 passes --- ||Best: 100.00 -- Worst: 100.00 -- Average: 100.00 || ||hope this helps || ||-kim || ||-- ||[EMAIL PROTECTED] ||___ ||Asterisk-Users mailing list ||Asterisk-Users@lists.digium.com ||http://lists.digium.com/mailman/listinfo/asterisk-users ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || || | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Adam Goryachev wrote: On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: 2) There isn't anything like what you want. I know, I want the same thing. There is no phone out there that will do this with any protocol that asterisk uses. This is the one major failing of asterisk ( and voip in general. I smell an oportunity for a phone manufacture ), and what keeps it out of a lot of places. It's alright, you can come out from under your rock now The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Yes, but each of those 6 lines on the 7960 must have their own seperate SIP username/password. And if you are a small office with 6 7960s, thats 36 username/passwords. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 license
Hi all. Dopes someone know how I can move a key license of the g729 codec from one to another machine? Find nothing usefull @ the wiki. Thnx 4 help in advance. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
The user name is the extension and the password is always the same. Not hard to configure. Yes, but each of those 6 lines on the 7960 must have their own seperate SIP username/password. And if you are a small office with 6 7960s, thats 36 username/passwords. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA2000 dialplan vs Asterisk dialplan
Trevor Peirce wrote: Steve Prior wrote: the SPA2000 does for me over the one in Asterisk. Is there a way to disable the use of the SPA2000 dialplan so I don't have to keep it in synch? Or is there some reason why it would be a bad idea for me to do so? Sure just put x. as your dial plan and any number will be accepted. The catch is you'll have to wait for the Short (Long?) Digit Timeout to pass before the call goes to asterisk for processing. If the SPA has an idea of what digit combinations are accepted it will wait until it has a match and send the call along at just the right time. No delays waiting the digit timer to expire. Thanks for the info. I also have an IAXy and it doesn't have anything like the concept of a dial plan in the ATA, but I've never noticed any kind of digit delay either - how long is this timeout on the Sipura? So does the IAXy just send digits as they come in whereas the Sipura tries to collect them and send them all at once? It sounds like a strategy might be to use the X. dialplan while I'm tinkering with the Asterisk side of things, and once I've got something I want to keep stable I'd make the Sipura dial plan more specific. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
Actually called Digium with this exact question last week. They said that you can register the new license on the new server provided that you ony registered it once before. They said there is no unregister script to unregister the license from the old server, however. If you have already used up your 2 registrations, you will need to contact Digium for assistance on this. I also asked if leaving the keys on my dev. box would cause a conflict (also was pretty clear that I wanted to be in compliance with their license agreement) and the lady said there was no problem and leaving the old keys on the dev. box would not cause a conflict. On 5/2/05, Peter [EMAIL PROTECTED] wrote: Hi all. Dopes someone know how I can move a key license of the g729 codec from one to another machine? Find nothing usefull @ the wiki. Thnx 4 help in advance. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
Yes same provess you did to register the license in the first place. You can rereg the license I think 3 times or so before you have to call Digum and have them manually change what your license is tied to. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice limits???
- Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Kerry Garrison' Sent: Sunday, May 01, 2005 2:50 AM Subject: RE: [Asterisk-Users] Broadvoice limits??? Broadvoice. Seems to be no limit on inbound, but I found any channels after 5 outbounds would get an immediate disco. Guess I'll have to stick to Vonage to blast into the local radio shows. Or maybe 5 on BV, 5 on Vonage, and X on the PRI. - Are you manually dialing out that many times or have you got some script to do it for you? Would be nice if there was a *66 feature (Automatic Callback Activation). ___ Mobilcom http://www.mobilcom.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 Forward problem codec issue
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests, but
it is doing it now. I will try to track down the specific change tonight ...
My solution for now is to Answer() the call before dialing out. I changed all
of my outbound dialing rules from:
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to the
remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing.
During the time that the caller should hear ringing the * console produces
pages of errors.
snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format g729 since our native format
has changed to ulaw Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read:
Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw ..
/snip
I have tested this with the phones behind a PIX firewall with NAT, behind a PIX
firewall without NAT, and without a firewall at all. Nat is not the problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the *
server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten = --6081,1,Dial(SIP/--6081SIP/--6091,20)
=SNIP
-- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) in new
stack
-- Goto (TPN,--6081,1)
-- Executing Dial(SIP/---.---.241.35-40400490,
SIP/--6081SIP/--6091|20) in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 Moved Temporarily back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268)
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490
-- SIP/---.---.241.35-f522 answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered SIP/---.---.---.35-40400490
== Spawn extension (TPN, --6081, 1) exited non-zero on 'Local/[EMAIL
PROTECTED],2ZOMBIE'
-- Attempting native bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==/SNIP
Now here is the console output with a single phone defined in the
extensions.conf (exten = --6081,1,Dial(SIP/--6091,20)
*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) in new
stack
-- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in new
stack
-- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to 'Local/[EMAIL PROTECTED]'
(thanks to SIP/--3263-f670)
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of
format
g729 since our native format has changed to ulaw pages of the same error
Apr 29 11:19:18 NOTICE[2185]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format
g729 since our native format has changed to ulaw
-- SIP/---.---.241.35-4e1f answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered
Re: [Asterisk-Users] g729 license
Hi. I'ved registered it for 2 times, so I've got to contact digium. Thnx 4 info. On Mon, May 02, 2005 at 10:26:35AM -0400, Pedro wrote: Actually called Digium with this exact question last week. They said that you can register the new license on the new server provided that you ony registered it once before. They said there is no unregister script to unregister the license from the old server, however. If you have already used up your 2 registrations, you will need to contact Digium for assistance on this. I also asked if leaving the keys on my dev. box would cause a conflict (also was pretty clear that I wanted to be in compliance with their license agreement) and the lady said there was no problem and leaving the old keys on the dev. box would not cause a conflict. On 5/2/05, Peter [EMAIL PROTECTED] wrote: Hi all. Dopes someone know how I can move a key license of the g729 codec from one to another machine? Find nothing usefull @ the wiki. Thnx 4 help in advance. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Chris Mason (Lists) wrote: The user name is the extension and the password is always the same. Not hard to configure. With the SNOM 220, you have five buttons/lamps that can be used as line appearances--these buttons can each register to a different SIP URL. Each sidecar has 20 buttons/lamps, and you may have up to three sidecars. Using the hint priority in Asterisk, the buttons serve as extension busy lamps. You can also use these buttons to transfer calls. I have an executive suites customer where each tenant is a separate business. For an incoming call, the attendant needs to know which DID number is being called so she can answer with the proper greeting. I would like the sidecar buttons to be able to register to a SIP URL, so an incoming call would blink the tenants button, but that is not possible--I can only use the five buttons on the phone for that purpose, and there are more than five tenants. A suggestion was to alter the Called ID Name to the DID number. This would work for the attendant, but the tenant would like to see the original Caller ID Name. I would rather not have to put a PC at the attendants position, but that is the way this is shaping up. Does anyone have any suggestions? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording calls
user monitor application -Wix ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pb SIP and port
Hi all, I try to dial via a Softswitch a number : exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf) And my sip.conf [SIP-OUT] type=peer host=10.XX.XX.XX defaultip=10.XX.XX.XX disallow=all allow=g729 allow=ulaw allow=alaw context=Ipnotic canreinvite=yes nat=yes dtmfmode=rfc2833 But it does'nt work... * try to dial with the port 5060 when i specify to him to dial on the 5061 one... Any idea ? Thx -- Guy Decarpentrie - ipnotic - switch to ip Responsable système Tel / Fax : 01.72.29.05.08 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ExtensionState problems using Asterisk API
Da: Aurelio Forese
Inviato: luned 2 maggio
2005 16.13
A:
'asterisk-users@lists.digium.com'
Oggetto: ExtensionState problems
using Asterisk API
Im trying to write a web application in php to
monitor the extension state of my asterisk peers. My application is working but
Im able to recognize only some status:
-1: when a peer is inexistent
0: when a peer is logged and ready for calls.
1: when a peer is busy in a call (it is indifferent if it calls or
receives the call)
3: (I dont
know what it is! In the documentation it means Digits or equivalent have
been dialled but this is
The default response that asterisk gives me when
a peer is existent but not logged).
4: The called peer is ringing
As you see asterisks response are a little bit
confusing me! Ive put hint priority to my all my peers in local
context in my manager.conf so I can receive the extensions state. Have
you some suggestions or can you solve the problem? Are there any particular
configuration parameters to put on my manager.conf? As an example I show you
how a sip is defined in my manager.conf
[local]
; Example NAT
exten = 805,hint,SIP/605
;exten = 805,1,Macro(local,SIP/605)
exten = 805,1,Dial(${Example NAT},60,Ttr)
exten = 805,2,SetLanguage(it)
exten = 805,3,Voicemail(405)
exten = 805,4,Hangup
exten = 805,102,GoTo(3)
exten = 805,103,Playtones(busy)
Please let me know! Thank You
___
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RE: [Asterisk-Users] g729 license
Here is how Digium license works. 1)You are allowed to register 2 times , for which digium license server does not object. 2)If you want to register it a third time on a different server, send an email to digium and they increment (or decrement the registrations, in the real sense) the number, so that you can register it again. Send an email to [EMAIL PROTECTED] with your license details and they will increment it for you. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Sent: Monday, May 02, 2005 10:31 AM To: Pedro; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] g729 license Hi. I'ved registered it for 2 times, so I've got to contact digium. Thnx 4 info. On Mon, May 02, 2005 at 10:26:35AM -0400, Pedro wrote: Actually called Digium with this exact question last week. They said that you can register the new license on the new server provided that you ony registered it once before. They said there is no unregister script to unregister the license from the old server, however. If you have already used up your 2 registrations, you will need to contact Digium for assistance on this. I also asked if leaving the keys on my dev. box would cause a conflict (also was pretty clear that I wanted to be in compliance with their license agreement) and the lady said there was no problem and leaving the old keys on the dev. box would not cause a conflict. On 5/2/05, Peter [EMAIL PROTECTED] wrote: Hi all. Dopes someone know how I can move a key license of the g729 codec from one to another machine? Find nothing usefull @ the wiki. Thnx 4 help in advance. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_sccp - status
The cisco 7960 works well with * and SIP. Out of curiosity I loaded the ccm version 7.1 and tested it briefly with CVS HEAD * and latest chan_sccp. The interface when using ccm load on the phone is certainly different. Things I don't see how to fix are: o Setting the date and time on the phone o The vm button makes a msg on * saying VM Button is not yet handled o When on a call there is no transfer button. This must be something the chan hast to tell it display Or not? o It seems like the * console is very busy with messages constantly on it. This likely means more processor power needed for large #s of these phones. Just a thot. Someone may have some real life experience. o I don't see any way of making * read changes to sccp.conf. Tried a * reload. And a module reload. But had to stop * completely to get it to reread the config change. o The phone wants DISTINCTIVERINGLIST.XML. What does that look like? Is anyone using them in real life? The wiki seems to have little information. Like how to setup the ring tone file, the locale etc. Thoughts? -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
On May 2, 2005 10:31 am, Charlie Watts wrote:
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests,
but it is doing it now. I will try to track down the specific change
tonight ...
My solution for now is to Answer() the call before dialing out. I changed
all of my outbound dialing rules from:
Same problem encountered here. My solution is to answer and play a sec of
silence before the dial proceeds - if i don't answer both parties are
connected but can't hear each other.
joe
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing. During
the time that the caller should hear ringing the * console produces pages
of errors. snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw Apr 30 08:41:03 NOTICE[2813]:
channel.c:1314 ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw .. /snip
I have tested this with the phones behind a PIX firewall with NAT, behind a
PIX firewall without NAT, and without a firewall at all. Nat is not the
problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the
* server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten =
--6081,1,Dial(SIP/--6081SIP/--6091,20)
=SNIP
-- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) in
new stack -- Goto (TPN,--6081,1)
-- Executing Dial(SIP/---.---.241.35-40400490,
SIP/--6081SIP/--6091|20) in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 Moved Temporarily back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490 -- SIP/---.---.241.35-f522 answered
Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered
SIP/---.---.---.35-40400490 == Spawn extension (TPN, --6081, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' -- Attempting native
bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==/SNIP
Now here is the console output with a single phone defined in the
extensions.conf (exten = --6081,1,Dial(SIP/--6091,20)
*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) in
new stack -- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in
new stack -- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/--3263-f670) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw pages of the same error Apr 29
[Asterisk-Users] OT: DSL problems - UNREACHABLE - REACHABLE
I could really use some input here, forgive the OT nature, but my problem is related to asterisk and voIP on a DSL connection and becoming a big mystery. I noticed about three weeks ago a lot of UNREACHABLEs that became REACHABLE 10 seconds later. After studying this a little, it happens that the DSL connection was stopping every 8 minutes (+ about 3 seconds). The modem doesn't apperat to lose sync, the data flow just stops. Since then I've removed asterisk from that connection. Every possible test has been done at our office, three different modem/routers of different brands were swapped in/out, there is a second phone line in the same cable that is on a different connection and it does not have the interruptions. I've turned off every box in the office and disconnected every cable from the router.Also disconnected FXO lines, phones and left just a modem/router on. No change. The 8 minutes are invariable, so after turning of everything here, I can't see how it could possibly be any local hardware. The phone company here has, after being evasive aboput checking the DSLAM, claimed they did everything possible, changed our DSLAM connection, tried every piece of equipment on their end. Ditto the ISP who has been very cooperative. I can only think of one more possible approach: get the power lines and the phone line independently checked for some kind of parasitic interference, say a big machine of some kind going on and off. Why this affects one DSL connection and not the other... I wouldn't know. Does anyone have any suggestions about what kind of outfit to look for that might do this kind of checking? Or any suggestions to pursue at all? tia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pb SIP and port
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guy Decarpentrie wrote: Hi all, I try to dial via a Softswitch a number : exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf) And my sip.conf [SIP-OUT] type=peer host=10.XX.XX.XX defaultip=10.XX.XX.XX disallow=all allow=g729 allow=ulaw allow=alaw context=Ipnotic canreinvite=yes nat=yes dtmfmode=rfc2833 But it does'nt work... * try to dial with the port 5060 when i specify to him to dial on the 5061 one... Any idea ? Try [SIP-OUT] type=peer host=10.XX.XX.XX port=5061 defaultip=10.XX.XX.XX HTH - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQnY/10tP/KMNOfRbAQJvaQf/VsJjPmnl4062UV8U9XB1Kkl8HvL6nk3t EqPAP04wPzCsqDO8FcgV1B4mqHCVh6hm4Wx5J2KPuUczRw2PzDRsirAyl1MMsYVE 9x6PfIm4HgNXa0yax5Am9J1ngrdbcRtVGGwyeqCNoNJtgHRdkbfiQ1TTEb+GOGmD zPqO8nMPMknBbj6Rp5rVlqF/m4F97whO9nTvO9n/A4KO3/a+7at+6sJd7VY6Hnlb Fb/kLCtqZQZeBFzKD75xJ4TWz84Yk5X2OCcLz9WTrCKVYrShrpOYyHFd0lMMON+Z QDZgMFF2Wsz8H8RqYQjwQxDGCrf7ABPD35AzuWVzPFWAwihQoaAAHw== =3QEA -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as VM for Nortel System
Hi, Can anyone think of a way to use asterisk as the voicemail system for a Nortel phone system? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Timeout
Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. Thank you for your help - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to cancel a transfer in progress:
Is there a feature code you can dial after beginning an atxfer (*2) that will bail out and return you to the caller. Let's say I want to transfer to the CEO of the company, but only if he is available. Once I hit *2, punch in his extension, I don't of anyway to cancel out. If I hit * or hangup, the transfer completes anyway. No other keys seem to do anything once the transfer has started. I saw one person asking the same thing in a comment on the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20features.co nf#comments Anyone have an answer, or does this need to be added? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] large scalable voip setup
Hi all. Is there anyone who have a big experience with large scalable voip setup and want to share some experience, knowlegde? I need to handle a lot concurrent calls, to pstn and to sip gateways' The current setup can't handle the load anymore. I've some solutions in mind, but don't know if it fits well. If someone is willing to communicate, it would be every appriciated. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback() stops working.
Simon Morris wrote: On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. I have heard from 5 or so people about this problem. I run CVS STABLE (almost the same as 1.0.7) and had none of these issues. I wish I could help you, but I wanted to let you know that this is not a general problem. Well a quick reinstall of Asterisk solved the problem, but I hope it doesn't happen again :) Thanks for the heads up Eric. I'm having the same problem. What exactly did you do yo reinstall it? Just a make make install? Thanks -Rob begin:vcard fn:Robert Derr n:Derr;Robert org:WeatherFlow, Inc.;IT Florida office adr:;;120 Canal St;New Smyrna Beach;FL;32168;USA email;internet:[EMAIL PROTECTED] title:Software Developer tel;work:386-423-1516 tel;fax:386-409-5178 url:http://www.iwindsurf.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as VM for Nortel System
On May 2, 2005 11:07 am, Matt wrote: Can anyone think of a way to use asterisk as the voicemail system for a Nortel phone system? use a couple ATAs or an 8 port ATA card and wire them up to FXO ports on *, have the extensions callforward-busy/unavail to the analog extensions. * can take the voicemail and hopefully get the number it redirected from so it can hookflash and *1[exten of asterisk] to toggle MWI. Not the best solution but an option. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DSL problems - UNREACHABLE - REACHABLE
On May 2, 2005 10:56 am, Wilson Pickett wrote: The phone company here has, after being evasive aboput checking the DSLAM, claimed they did everything possible, changed our DSLAM connection, tried every piece of equipment on their end. Ditto the ISP who has been very cooperative. Can you get stats out of the DSL modem as far as retries and whatnot? I get that occassionally on ours as well (Sangoma S518 ADSL PCI modem) but the modem isn't showing anything bad (the odd collision but that's not 10s long) -- basically it's something deeper and I believe on the DSLAM side. I can only think of one more possible approach: get the power lines and the phone line independently checked for some kind of parasitic interference, say a big machine of some kind going on and off. Why this affects one DSL connection and not the other... I wouldn't know. The fact that the link doesn't drop suggests that's not the problem but I can't be 100% sure. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, h323
Hello I've installed asterisk with [EMAIL PROTECTED] package with h323 support. I've a Digium TDM10B card and we have a quintum voip gateway. I'm trying to make call with an analog phone plugged to that card through our quintum with h323 protocol. How to confgure related files? Any help welcome. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please find me a IAX provider
Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Timeout
On Monday 02 May 2005 16:07, Dan Levine wrote: Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. http://www.voip-info.org/wiki-asterisk+manager+events Connect to the Manager interface as part of the PPP script executed when you get a new IP address, and then issue an Event: Reload gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pb SIP and port
Le lundi 2 Mai 2005 16:57, Ron Wellsted a écrit : Guy Decarpentrie wrote: Hi all, I try to dial via a Softswitch a number : exten = 01x,3,Dial(SIP/[EMAIL PROTECTED]:5061,30,trf) And my sip.conf [SIP-OUT] type=peer host=10.XX.XX.XX defaultip=10.XX.XX.XX disallow=all allow=g729 allow=ulaw allow=alaw context=Ipnotic canreinvite=yes nat=yes dtmfmode=rfc2833 But it does'nt work... * try to dial with the port 5060 when i specify to him to dial on the 5061 one... Any idea ? Try [SIP-OUT] type=peer host=10.XX.XX.XX port=5061 defaultip=10.XX.XX.XX HTH Great ! Many thanks. ++ -- Guy Decarpentrie - ipnotic - switch to ip Responsable système Tel / Fax : 01.72.29.05.08 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: DSL problems - UNREACHABLE - REACHABLE
Welcome to DSL, the telco didn't do any more tests then required to get sync for 30 seconds. Cancel the DSL and get another line. That's about the extent of it, or at least in Ontario it is, I've had this problem with 5 or 6 connections. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: May 2, 2005 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: DSL problems - UNREACHABLE - REACHABLE I could really use some input here, forgive the OT nature, but my problem is related to asterisk and voIP on a DSL connection and becoming a big mystery. I noticed about three weeks ago a lot of UNREACHABLEs that became REACHABLE 10 seconds later. After studying this a little, it happens that the DSL connection was stopping every 8 minutes (+ about 3 seconds). The modem doesn't apperat to lose sync, the data flow just stops. Since then I've removed asterisk from that connection. Every possible test has been done at our office, three different modem/routers of different brands were swapped in/out, there is a second phone line in the same cable that is on a different connection and it does not have the interruptions. I've turned off every box in the office and disconnected every cable from the router.Also disconnected FXO lines, phones and left just a modem/router on. No change. The 8 minutes are invariable, so after turning of everything here, I can't see how it could possibly be any local hardware. The phone company here has, after being evasive aboput checking the DSLAM, claimed they did everything possible, changed our DSLAM connection, tried every piece of equipment on their end. Ditto the ISP who has been very cooperative. I can only think of one more possible approach: get the power lines and the phone line independently checked for some kind of parasitic interference, say a big machine of some kind going on and off. Why this affects one DSL connection and not the other... I wouldn't know. Does anyone have any suggestions about what kind of outfit to look for that might do this kind of checking? Or any suggestions to pursue at all? tia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 codec order
Hello, Is there a particular order in which codec should be entered in the oh323.conf file? I believe that they are put in order of priority. But depending on which codec is put before another, even if the caller does not support all of them. Let me clarify. I have a cisco ATA. When I have this in my oh323.conf file, codec=G7231A6K3 frames=1 codec=G729 frames=1 codec=GSM0610 frames=1 codec=G711A frames=1 codec=G711U frames=1 With the G711 codec i cannot hear any sound. In the debug, it shows that g711 is used. But when I have. Other codecs(g723 and g729) seems to work fine. codec=G711A frames=1 codec=G711U frames=1 codec=GSM0610 frames=1 codec=G7231A6K3 frames=1 codec=G729 frames=1 I can hear the a very very slowed down robotic voice. Thanks Kido ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Sip TEXT Messaging
So I went out and got this IP 500 phone, and see that it has something called SIP Text messaging. I can find NO DOCUMENTS out there in Internetland referring to how this works, or any utility to send it messages. Id love to be able to send reminders and such to a phone, or group of phones. Is this possible? Or is this a Polycom-only feature? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
On Mon, 2005-05-02 at 09:02 -0500, Matthew Boehm wrote: Adam Goryachev wrote: The polycom ip600 and cisco 7960 both have 6 lines available. Yes, but each of those 6 lines on the 7960 must have their own seperate SIP username/password. And if you are a small office with 6 7960s, thats 36 username/passwords. So? ?? With 100 of them, you have 600 entries in sip.conf ... I don't see the problem? Just write some simple script to create the entries automatically... I wrote something like a 20 line shell script to build the polycom xml files, and put them in the FTP dir, add the entries to the sip.conf, and also add the entries for the extensions.conf (actually, I used the asterisk DB magic, but same thing)... So, to provision a new phone, I just: ./newphone macaddress extension passwd and it is all done... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
On Mon, 2005-05-02 at 08:40 -0600, Michael Welter wrote: A suggestion was to alter the Called ID Name to the DID number. This would work for the attendant, but the tenant would like to see the original Caller ID Name. Is there an original caller id name ?? You *might* be able to setup some dialplan magic to re-write the CIDName again when transferring it. ie, overwrite the change you made when you added the tennant name in front... I would rather not have to put a PC at the attendants position, but that is the way this is shaping up. Does anyone have any suggestions? Yeah, this sounds like the next best thing, while it may not sound like a nice thing, I've had various thoughts about how 'cool' this could be. Consider placing a web front end to allow your tennants to update their greeting at will. When the call comes in, the 'current' text is displayed on screen for your receptionist to say. Lotsa other things are possible once you get to this stage... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please find me a IAX provider
Kumara Jayaweera wrote: Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara Well, I dig voicepulse, but I don't know what kind of latency you'd be running into. Check out: connect.voicepulse.com. I don't know what the proper etiquette is regarding their iax2 server addresses, so if you want them to ping them, I'd ask them. I'm sure they'd give them to you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug found in SJLabs SJPhone concerning dialpad
Seems as though the dialpad in SJPhone cannot me used to signal *. *2 doesn't do anything except play a DTMF in your ear. If you use your keyboard to send shift-8, 2, all works as expected. Bug report submitted already. Cheers Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Timeout
The Box Itself doesn't get a new IP address, the router does. What I'm looking to do is have the IAX connection re-register every hour or so. Is this the right idea? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Monday, May 02, 2005 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IAX Timeout On Monday 02 May 2005 16:07, Dan Levine wrote: Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. http://www.voip-info.org/wiki-asterisk+manager+events Connect to the Manager interface as part of the PPP script executed when you get a new IP address, and then issue an Event: Reload gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please find me a IAX provider
Voicepulse is great... - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Monday, May 02, 2005 11:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please find me a IAX provider Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please find me a IAX provider
Kumara Jayaweera wrote: Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. With 128 kbps, you won't make more than 2-3 calls simultaneously... Best Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pr-pay! Essayez gratuitement - 5 crdits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please find me a IAX provider
Kumara Jayaweera wrote: Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. With 128 kbps, you won't make more than 5-6 calls simultaneously... Best Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pr-pay! Essayez gratuitement - 5 crdits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio cut off at beginning of call
On May 1, 2005, at 11:39 AM, Gene Naden wrote: When we call out from our Asterisk system we consistenly lose the first roughly 1500 milliseconds of the audio from the destination. This is easiest to demonstrate with a recorded announcement. In other words, Hello for example is missing. We are calling over the PSTN via a voice T1 line. We are using the stable cvs from about April 1. I searched lists.digium.com but did not find anyone with this problem using the PSTN. Does anyone have any ideas? Same here, via VoIP. I reported it to the list a while back: http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088514.html If you're getting it via ZAP and I'm getting it via VoIP, sorta starting to sound like a setup issue on the Asterisk side, doesn't it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk CDR Bug Or Not?
Good Day list, It appears that the CDR is inaccurate, (or I am inaccurate when reading it) when an attended transfer is conducted with a phones transfer button Example +-+++--- ---+---+---+---+--+- -+--+-+-+--+ -++---+ | calldate| clid | src| dst | dcontext | channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | uniqueid | userfield | +-+++--- ---+---+---+---+--+- -+--+-+-+--+ -++---+ | 2005-05-02 12:50:32 | ron hartmann 2004 | 2004 | 2097 | from-internal | SIP/2004-8835 | SIP/2096-73a9 | ResetCDR | w|0 | 33 | ANSWERED|3 | | 1115052597.220 | | | 2005-05-02 12:50:22 | ron hartmann 2004 | 2004 | 2097 | from-internal | SIP/2096-2ee4 | SIP/2097-7d9f | ResetCDR | w| 10 | 8 | ANSWERED|3 | | 1115052622.223 | I initially called extension 2096 from my extension 2004 I then Attended transfer to extension 2097. The problem is that if I look under dst, it shows both calls as extension 2097 (which is the final extension). Is this a bug, or desired effect? I would think it should look like the following +-+++--- ---+---+---+---+--+- -+--+-+-+--+ -++---+ | calldate| clid | src| dst | dcontext | channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | uniqueid | userfield | +-+++--- ---+---+---+---+--+- -+--+-+-+--+ -++---+ | 2005-05-02 12:50:32 | ron hartmann 2004 | 2004 | 2096 | from-internal | SIP/2004-8835 | SIP/2096-73a9 | ResetCDR | w|0 | 33 | ANSWERED|3 | | 1115052597.220 | | | 2005-05-02 12:50:22 | ron hartmann 2004 | 2004 | 2097 | from-internal | SIP/2096-2ee4 | SIP/2097-7d9f | ResetCDR | w| 10 | 8 | ANSWERED|3 | | 1115052622.223 | Thanks ~ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as VM for Nortel System
What type of Nortel system? Is it an option or a norstar? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 10:16 AM Subject: Re: [Asterisk-Users] Asterisk as VM for Nortel System On May 2, 2005 11:07 am, Matt wrote: Can anyone think of a way to use asterisk as the voicemail system for a Nortel phone system? use a couple ATAs or an 8 port ATA card and wire them up to FXO ports on *, have the extensions callforward-busy/unavail to the analog extensions. * can take the voicemail and hopefully get the number it redirected from so it can hookflash and *1[exten of asterisk] to toggle MWI. Not the best solution but an option. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing calls, X100P
I can't seem to be able to make outgoing calls with X100P card. I can receive calls fine and it picks up the line and sends the tones, but the telco doesn't recognize them. While the tones are sent I continue to hear the dial tone on the line when I pick up a parallel. I also cannot dial from the parallel until X100P hangs up the line. I am in Turkey. I imagine this is due to incorrect zone information, but I can't seem to be able to find the correct values for Turkey. I tried guessing them with no luck. Any help would be appreciated. Thanks -- Mehmet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Sound on PSTN End
I have the same problem on a Dell 1850 with a TE410P and have been attempting to narrow it down. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please find me a IAX provider
With 128 kbps, you won't make more than 2-3 calls simultaneously... Ignore this, I read 64kbps. I have corrected this in a follow-up message... /ducks -- Ykoz Un Max - La VoIP en pr-pay! Essayez gratuitement - 5 crdits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newer Dell Servers + TDM card
-Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Has anyone ever been able to fix this NMI power issue that the Dell's have with the TDM cards? Basically locks the machine up when trying to bring up the module. I get an NMI the first time I load the module, but the machine always recovers. Subsequent load/unload cycles don't trigger further NMIs. I'd like to know of any way to fix it, too, 'cause that orange flashing light is kind of annoying. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please find me a IAX provider
In all probabilty you will be able to make just one simulatneous call with that bandwidth, where you need two channels of 64 Kbps each in the two directions, using Ulaw ( assuming both users are blabbering at the same time). You don't need any IAX service providers. You just need a $10 Account with any SIP based VOIP Calling Card Company. I suggest that you sign up as a user for sip account at http://www.terracall.com This is simple and best. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Monday, May 02, 2005 11:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please find me a IAX provider Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing calls, X100P
On Mon, May 02, 2005 at 01:02:34PM -0400, Mehmet Tolga Avcioglu wrote: I can't seem to be able to make outgoing calls with X100P card. I can receive calls fine and it picks up the line and sends the tones, but the telco doesn't recognize them. While the tones are sent I continue to hear the dial tone on the line when I pick up a parallel. I also cannot dial from the parallel until X100P hangs up the line. I had the same trouble, here in the UK. What do you have rxgain and txgain set at in zapata.conf ? I found I had to raise the txgain to 0.25 to get it to work on my British Telecom line (Cable Provider's line was fine) I also had to up even further (to 3.0) in order to get inbound fax to work, although I think I can back that off a little. Of course, increasing the gain also gives me an echo issue, in proportion to the gain values, so I suspect I have an impedance issue. YMMV of course. Hope that Helps Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End I have the exact setup you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You might turn on qualify just to see if the * is seeing network flaws. Keep in mind, if your using windows, anytime the user starts clicking around, you can expect less than ideal audio. Also, why disable GSM ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim ChandlerSent: Monday, May 02, 2005 11:23 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
