Re: [Asterisk-Users] Trouble with call parking/transfer
Tim Pushor wrote: I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. Before I go ripping things apart, I was really wondering if this is by design, or should it work on all my devices? I have an iaxy, phones hanging off fxs ports on a pair of tdm400p's, a sipura 841, a sipura 3000, and a pair of sipura 2000's and a Polycom IP 500. It only works on the phones hanging off the tdm400p. Should this work on all phones? Does anyone have it working on non digium FXS phones? Sounds to me like you have a DTMF problem. Does other DTMF work from your non-Zap devices? Not dialing, I mean like an IVR or VoicemailMain. ___ 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] No busy tone when dialing out over ISDN with Polycom 500 IP
Kib Eki wrote: Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Try priindication = inband in /etc/asterisk/zapata.con ___ 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] ignorepat doesn't work
Jerry wrote: The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. Grandstream BT phones don't have a digitmap feature. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Best of the best of IP Phones
Greg Boehnlein wrote: I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom Soundpoint IP-500 and 600 to my Cisco's now. All things being equal between the phones, the following are why I prefer the Polycoms: 1. Better speakerphone than the Cisco 7960s. Despite the fact that Cisco licensed Polycom's SpeakerPhone technology, the SoundPoint IP 500 and 600 just sound and work better. 2. Lower price point: $185 for a NEW SoundPoint IP 500 is better than the $225 I see for used 7960s. 3. FTP based provisioning. TFTP is fine, but doesn't work very well through some NAT implementations. The PolyCom's can be centrally provisioned from any FTP server, and NAT doesn't seem to be a problem for it. 4. More intuitive User Interface. My clients require less training and are up and running quicker on the Polycoms. These are my opinions I love BOTH phones, and you can't go wrong with either choice, but for my needs the Polcom's work a lot better. The Polycoms also include a power supply and SIP firmware, which the Ciscos do not. Overall I just think the Polycoms are a better value. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] ignorepat doesn't work
Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] ztcfg doesn't do anything from /etc/rc.d/rc.local
Chris wrote: You need this before wcfxs /sbin/modprobe zaptel *sigh* zaptel will automatically load when the card driver loads. modporbe will also run ztcfg after loading the card driver because (if you ran make install) /etc/modules.conf tells it to do so. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Best of the best of IP Phones
Ron Wellsted wrote: I have to agree, the Cisco 7960 is probably the best (I have yet to try a 7970/71). Cisco are a pain to deal with (they only want to deal with large value customers/distributors) and the phone do have some small quirks/bugs but they are the best in functionality and build quality. They are also the best speaker phone for small conferences. Have you tried Polycom? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Line Noise UPDATE - If you've got line noise, read this
Paul wrote: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I use X100P in at least several different Asterisk system and have no problems. One of them is an Intel motherboard, one of them is a Supermicro, one of them is ASUS motherboard, one of them is an older Compaq system. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] NAT issues
Steven Langley wrote: I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites the sip message with the IP of the Nat and the external port. It still works, but only if there is a constant flow of rtp traffic. If there is a break in the traffic, then the connection is lost. However, this problem may be to do with the fact that pinging is disabled on our network, but not sure. If you are the same person I spoke with on IRC then you forgot to mention that the SIP clients use VAD and that it cannot be disabled. ___ 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 and VAD
Pavel Siderov wrote: Is it possible turn on/off VAD (silence suspression) w/ Asterisk? Asterisk does not support VAD so it doesn't make sense to be able to disable 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
Re: [Asterisk-Users] NAT and only been able to have 1 SIP phone behind
Anton Krall wrote: Guys. Ive read on the wiki that a common problem with nat is that you can only have 1 sip phone behind, how do you get around this issue? Having a sip enabled router behind the nat like the GS 488 489 or 486? Or how have you done it without having any kind of linux box (SER or *) behind the nat. The idea is to have 2 or more sip clients behind some NAT and been able to connect to a remote asterisk box. I don't know why people think this. Any router with PAT (Port Address Translation) should work with multiple SIP clients behind NAT. Most routers support PAT (but may not call it that). ___ 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] Billing
Rizwan Chaudhry wrote:
Hey
I want to implement billing in Asterisk for a calling card type application.
My scenario is like this: PSTN = Asterisk =(IAX)= Asterisk = PSTN.
I used the ${ANSWEREDTIME} and ${DIALEDTIME} variables but
${ANSWEREDTIME} always gives a value even if the call is not answered.
e.g. If I dial on a Zap Channel, Zap answers the call the moment the
channel starts ringing. So I get an answeredtime even if there has
only been ringing.
Has anyone encountered this before?
This is the way it works with ANALOG FXO ports.
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Re: [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???
Damian Funnell wrote: When I asked them for further information on how to improve this they replied: ** Extract begins ** SCSI RAID can cause the problem. If disabling hyper threading does not resolve your problem my next suggest would be to revert to a PATA IDE hard drive solution configured to UDMA level 2 using hdparm. SCSI or SATA causes problems on some systems from what I have seen. The problem increases when using a SCSI or SATA RAID. ** Extract ends ** I really hope that they are wrong, as I don't feel like throwing away my nice expensive Ultra320 SCSI RAID controller and hot plug drives and replacing them with some crusty old IDE config. Needless to say I'm not going to go and shell out on IDE controller drives until I'm a little more certain that this is actually a problem and have asked them for more information. I've gotten some iffy advice from Digium tech support before. There is not a specific issue with RAID that I know of. However, it is common for some kernel modules to lock interrupts for very long amounts of time. This will cause problems. Graphics is well known for this, as is some RAID drivers. ___ 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] OutBOund Dial problem
kurt x wrote: I have the following extension (7700) that can dial out with the below config. exten = _1nxxnxx/7700,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/7700,2,Hangup If I change it to exten = _1nxxnxx/77XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/77XX,2,Hangup It does not work. It won't. Try: exten = _1nxxnxx/_77XX,1,Dial(SIP/[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] Server failing to Boot with
Brian Watters wrote: We have a Dell 1550 server and find that when attempting to start the server with any one of the four new Digium Wildcard X100P OEM FXO PCI cards the server will not even power up much less boot, upon removing the PCI card it will boot no issues, Placing any other PCI card in the server it boots no issues .. It appears to just plain not like these new cards .. Any ideas what to look for and or how to overcome this issue?? Someone else posted a message with the exact same problem earlier today. I didn't see any responses. ___ 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 Revision question.
So the only thing you have not done is tried the cards in a different system with a different motherboard. It is WELL KNOWN that the cards will not work well if they are shareing interrupts with another device. Ian Pattison wrote: I don't know how everyone else is doing but my woes are continuing. Hardware: Digium TDM400P (REV G according to the silk screening on the board) 2xFX0, 2xFXS purchased in August/September 2004 Dell Precision 420 (PIII-733, 512MB RAM nothing fancy but not doing too much either) Software: Zaptel, Libpri and Asterisk (v1-0) downloaded and re-compiled from CVS today (April 17) SuSE 9.1 (Kernel 2.6.4-52-default) configured as a life-support system for Asterisk only... no other apps running. Here's are my issues: 1. dmesg reports the card as Revision E/F although Rev G visually confirmed (see below) Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device :03:05.0 PCI: Sharing IRQ 11 with :00:1f.3 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) 2. Low ringing voltage still (~44V AC). I have used the boostringer=1 option when loading wcfxs, did I miss something at compile time? 3. Rogue on-hook 109V AC voltage (11V AC off-hook) on both FXS ports. I have conformed that it is being generated by the card itself. I repeat, it is not being induced on the wire. After finding it a the wall jack I was able to sample the same 109V AC at the card itself with no cables attached. 4. Random calls dropped on the FXO ports from both FXS and SIP clients. The drop is usually preceded by a 2-3 second buzzing sound on the line. This occurs with both incoming and outgoing calls. It should be noted that the card is sharing an IRQ with another device (the USB controller to be exact). No matter what slot the card is inserted in it ends up sharing an IRQ. To that end I made sure it was sharing with an unused device (no USB devices attached). Looking for help here... Thanks, Ian ___ 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] MeetMe
Matt Schwartz wrote: Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? MeetMe requires Zaptel. If you do not have Zaptel installed, MeetMe won't build. ___ 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 Revision question.
Rich Adamson wrote: My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. Doesn't anyone use Google anymore? http://lists.digium.com/pipermail/asterisk-users/2005-April/098934.html Also: http://www.google.com/search?hl=enq=site%3Alists.digium.com+boostringerbtnG=Google+Search -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Analogue phone transfering
David Wilson wrote: Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the flash key. I don't seem to get another dialtone as indicated in: http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer Any ideas what I've done wrong ? This is my zapata.conf: [channels] ; For analogue phone signalling=fxo_ks context=default channel=4 relaxdtmf=yes threewaycalling=yes transfer=yes adsi=no usecallerid=no rxgain=70.0 txgain=50.0 In zapata.conf you set options and then APPLY the options to a channel. As you can see you are specifying the channel before most of your otions so they are never applied. Move your channel= line AFTER the options you want to set. You might want to remove your rxgain and txgain so you don't blow out your eardrums. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] *8 nor *8# works for me!
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Friday, April 15, 2005 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] *8 nor *8# works for me! I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 Revision question.
Rich Adamson wrote: My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. Doesn't anyone use Google anymore? http://lists.digium.com/pipermail/asterisk-users/2005-April/098934.html Also: http://www.google.com/search?hl=enq=site%3Alists.digium.com+boostringerbtnG=Google+Search Eric, those links have nothing to do with his stated problem. The problem is 105v AC on the pstn line when on-hook and no ringing. The first line of this message says My specific issue has to do with ringing on my FXS ports. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 became berserk when Internet connectionis down and can't register to SIP server.
Andre Normandin wrote: The same thing happened to me a few days ago.. Truthfully, I thought it was just me, and a coincidence.. My DSL line went down, and astertisk refused to work until it came back up... I couldn't even dial out, nor would it receive calls on my 3 analog (X101P card) lines I don't know about anyone else, but if I decide to offer * to my clients, and tell them that your internal phones will go down if your internet blips, I think I'll be thrown out and the door locked behind me.. What's more, I don't feel that any programming should be needed on my end to have asterisk just silently mark any connection as unreachable just as it does now to sip clients when they do not register or respond to a qualify=yes directive.. Linux doesn't hang if it cannot resolve a dns, why should it make * go crazy? There is an attempted fix in CVS-HEAD (dnsmgr?). It's a known problem. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Why does this Macro Loop?
Mystery Glitch wrote:
In my [incoming] context I have something like this:
exten = 8885861575,1,Macro(vrforward,${EXTEN},8136361451)
And thie Macro contains this:
[macro-vrforward]
exten = s,1,GotoIF($[${CALLERIDNUM} = 954555]?40:2)
exten = s,2,SetGroup(${ARG1})
exten = s,3,CheckGroup(3)
exten = s,4,SetAccount(${ARG1})
exten = s,5,Dial(SIP/[EMAIL PROTECTED],30,o)
exten = s,40,AGI(checkin|${ARG1})
exten = s,41,Hangup
Change exten = s,40,AGI(checkin|${ARG1}) to exten = s,40,Noop
If it stops looping then the checkin script is causing the loop.
Perhaps it's running the macro-vmforward from within the script.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] RTP problem
trixter http://www.0xdecafbad.com wrote: I have done some further research, the first RTP packet is sent when playback() is called. No others. The application is running, if I press a key and goto a different item that would cause a new playback()/background() 1 more RTP packet is sent. To be clear If I call myself, RTP packets are sent. During a wait no packets are sent, only when playback() starts, and then only the 1 packet. This is true of any sip phone I have tried, whether or not it is local or remote. I can also call out through asterisk and that works, it just appears to be having a problem sending packets if it has to create the noise. As such this makes asterisk less than usable in my given situation. Is Asterisk getting a stream of RTP packets from the SIP client? What happens if you start talking on the SIP device? Does Asterisk then start sending RTP? It still sounds like VAD and silence supression is enabled on the SIP device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] BOUNTY - ztdummy modules
This message is to announce a bounty for the following: If ztdummy is already loaded, generate an error to the console and syslog when modules for Digium cards are loaded. If a modules for a Digium card are already loaded, generate an error to the console and syslog when ztdummy is loaded. You should use ztdummy OR a Digium card driver, but not both. Bounty is US$25 and will only be paid via PayPal. The patch must be accepted into Asterisk CVS-HEAD. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] BOUNTY: app_hangup from exten = h
This is a bounty for a patch to app_hangup.c to generate an error when Hangup is called from exten = h. You should not call Hangup from exten = h. The bounty is US$10 and will be paid via Paypal. The patch must be accepted into CVS-HEAD before the bounty will be paid. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] BOUNTY: app_hangup from exten = h
Andrew Kohlsmith wrote: On April 14, 2005 08:31 am, Eric Wieling wrote: This is a bounty for a patch to app_hangup.c to generate an error when Hangup is called from exten = h. You should not call Hangup from exten = h. I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure the dialplan stops. If you do post-hangup processing that has some branching it's far simpler to simply Hangup at the various branches than to Goto(h,end,1). A lot neater, too. A warning perhaps, but it should not error out. exten = h will not be called unless the channel has ALREADY hung up. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] BOUNTY: app_hangup from exten = h
Andrew Kohlsmith wrote: On April 14, 2005 09:42 am, Eric Wieling wrote: exten = h will not be called unless the channel has ALREADY hung up. I understand that, which is why I'm still suggesting a WARNING and not an error. Something like No need to execute Hangup from the h exten, line is already hung up I would consider a NOTICE or WARNING OK. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Zap won't dial out?
No. Dial(Zap/1/) will dial out ONLY on channel 1 of the T-1. Tim Connolly wrote: Could this be caused by using dial commands like dial(ZAP/1/) instead of using ZAP/g1/x I assumed if you have only one T1, the Zap/1 and Zap/g1 were the same. Is this correct? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Thursday, April 14, 2005 9:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Zap won't dial out? Apr 14 08:52:44 NOTICE[13947] app_dial.c: Unable to create channel of type 'ZAP' (cause 0) Apr 14 08:56:52 NOTICE[13947] app_dial.c: Unable to create channel of type 'ZAP' (cause 0) I just started seeing this anytime I try to call out on my TE110XP. I can still receive calls, but no outgoing calls work. Using RHES4 w/Asterisk CVS-HEAD-04/11/05-22:51:51. A 'restart now' fixes it, but I can't find the reason for it. Any idea what would cause this? ___ 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 -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Call Parking timming out to the wrong extension
I'm able to call park just fine, I can pick up a call just fine. but if nobody picks up the call and Asterisk tries to send the call back to te extension that parks it, it fails. HELP! 001 -- Executing NoOp(SIP/0004f201e463-a-7650, EXTEN=3599 CONTEXT=toll-access) in new stack 002 -- Executing Dial(SIP/0004f201e463-a-7650, SIP/0004f200cf85-a) in new stack 003 -- Called 0004f200cf85-a 004 -- SIP/0004f200cf85-a-de7a is ringing 005 -- SIP/0004f200cf85-a-de7a answered SIP/0004f201e463-a-7650 006 -- Attempting native bridge of SIP/0004f201e463-a-7650 and SIP/0004f200cf85-a-de7a 007 -- Started music on hold, class 'default', on SIP/0004f201e463-a-7650 008 -- Executing Park(SIP/0004f200cf85-a-bbca, ) in new stack 009 == Parked SIP/0004f200cf85-a-bbca on 3516. Will timeout back to toll-access,s,1 in 30 seconds 010 -- Playing 'digits/3' (language 'en') 011 -- Playing 'digits/5' (language 'en') 012 -- Playing 'digits/1' (language 'en') 013 -- Playing 'digits/6' (language 'en') 014 -- Added extension '3516' priority 1 to parkedcalls 015 -- Started music on hold, class 'default', on SIP/0004f200cf85-a-bbca 016 == Spawn extension (toll-access, s, 1) exited KEEPALIVE on 'SIP/0004f200cf85-a-bbca' 017 -- Stopped music on hold on SIP/0004f200cf85-a-bbca 018 -- Stopped music on hold on SIP/0004f201e463-a-7650 019 -- Started music on hold, class 'default', on SIP/0004f201e463-a-7650 020 == Spawn extension (toll-access, 3599, 2) exited non-zero on 'SIP/0004f200cf85-a-bbcaZOMBIE' 021 == Timeout for SIP/0004f201e463-a-7650 parked on 3516. Returning to toll-access,s,1 022 -- Stopped music on hold on SIP/0004f201e463-a-7650 023 == Starting SIP/0004f201e463-a-7650 at toll-access,s,1 failed so falling back to exten 's' 024 == Starting SIP/0004f201e463-a-7650 at toll-access,s,1 still failed so falling back to context 'default' -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. We set the NTP server and timezone using ISC DHCPd. option ntp-servers 172.16.7.1; option time-offset -21600; -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Invalid extension handling
Adam Robins wrote:
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten = i,1,Playback,invalid
exten = i,n,Goto(incoming,_NXXNXX,1)
;
[incoming]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Background(welcome); play welcome msg
ask for extension
exten = _NXXNXX,n,WaitExten(5) ; Wait for extension
This works fine, however, there is one special case that I would like to
handle differently. If the caller inadvertently presses the # key
following the extension, I would like to discard the # and then send the
call back onto the stack. I know how to strip the #, but I can't find
another command like WaitExten that will reprocess the call as new.
Use Goto. Since you have a pattern of _2XXX if they dial 2XXX# then
Asteirsk will process the call as 2XXX and just discard the # since
it's not listening for DTMF since it's already hit the Dial.
Same for _NXXNXX.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] PRI Advice...
Michael Crozier wrote: On Monday 11 April 2005 12:04 pm, Michael Crozier wrote: The zaptel drivers are proving quite unstable with this combination. If I attempt to rmmod the zap drivers, the machine hangs and is unresponsive to keyboard input, ping, or sysreq. Additionally, I attempted to bypass the Adtran unit using the TE110's to separate the data channels into an HDLC interface. I wasn't succesfull, as the HDLC interface triggered similar hanging problems and I never succeeded in getting packets of the data channels. For what it's worth, I changed to CVS HEAD and the instability problems modprobe'ing the modules seem solved. ifconfig down'ing an HDLC interface still produce a stack trace and a kernel thread hang, but it is at least sysreq sync'able. My HDLC Abort problems are lessened, but not solved. We haven't seen the problem in nearly five hours, reaching a new record. The HDLC problems were NOT solved or lessened. However, the telco (after insisting that they had throughly tested the lines) discovered that there was indeed a problem (unbalanced voltages in one of the pairs). When they fixed this problem, my HDLC Abort problem went away completely. The lesson I've learned: It's always the telco's fault :-) MOST of the time, in my experience, HDLC Abort problems are lost data caused by something locking interrupts for too long. As you experienced, it can also be caused by line problems. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] VAD/DTX implementation through zaptel cards
The only time PLC makes sense is thwn you are converting FROM VoIP to something else. So PLC would be done on chan_sip or chan_IAX, or chan_h323 on the receiving end. This is for 1.0.x. For CVS-HEAD you would want to do this on the receiving side in the PLC stuff. parijat wrote: Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] CVS-HEAD Zaptel with 1.0.x CVS Asterisk
Digium support suggested today that I run CVS-HEAD zaptel with 1.0.x CVS Asterisk. This seems totally wrong to me. Can others confirm? --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] PRI Errors with TE110P
Aaron Mathews wrote: I'm having a problem with a new digium te110p card. I'm running it on a T1 with PRI signalling, and everything works fine *except* I get errors every few minutes that look like the following: Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on 40 failed: Unknown error 500 Apr 11 23:23:04 NOTICE[10251]: chan_zap.c:6708 pri_dchannel: PRI got event: 8 on span 1 Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on 40 failed: Unknown error 500 Apr 11 23:23:04 NOTICE[10251]: chan_zap.c:6708 pri_dchannel: PRI got event: 6 on span 1 And continue on and on just like that. I found some old mailing lists posts from the beginning of 2004 that seemed to indicate that this was a 'frame buffering' problem, and that digium was working on a fix- is this still the case? Is there a fix? Something is locking interrupts on your system for so long that the Digium card is losing data from the PRI. It could just be a crappy motherboard (the SuperMicro board I got recently did this). Usually it's caused by the IDE and you can use the various things listed in the mailing list archives like unmasking interrupts, enabling DMA, etc to reduce the time interrupts are locked. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Pretty Voicemail Docs
Has anyone written up pretty voicemail user docs? I think voicemail is so easy even my cat can use it. However, my users are complaining about lack of docs for voicemail. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] VAD/DTX implementation through zaptel cards
No. parijat wrote: Pls could u be more elaborate as I am new to asterisk.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, April 13, 2005 7:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards The only time PLC makes sense is thwn you are converting FROM VoIP to something else. So PLC would be done on chan_sip or chan_IAX, or chan_h323 on the receiving end. This is for 1.0.x. For CVS-HEAD you would want to do this on the receiving side in the PLC stuff. parijat wrote: Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] 3-Way Calling in Asterisk
Wiley Siler wrote: As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference This is BS. 3-Way calling is supported on both Cisco and SIPura ATAs, using FLASH just like any other analog 3-way call. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] multiple line usage on Polycom IP300
You cannot disable call waiting on the polycoms. Therefore you need to use SetGroup and CheckGroup to keep Asterisk from sending more than one call to the same SIP peer at the same time. The polycom will ALWAYS accept a second call on a line that's in use. Wiley Siler wrote: If you have two lines registered to one phone then you need to do the following... This assumes extensions 1001 and 1002 are your line appearances... exten = 1001,1,Dial(1001,20,trf) ;we are dialing line 1 -- After 20 seconds it will timeout and go to the next line exten = 1001,2,Dial(1002,20,trf) ;just told it to dial line 1002 exten = 1001,3,Do your voice Mail Here exten = 1001,4,Hangup You could alternately just use a GoTo after the 1st dial attempt times out and send the call to 1002 If you are talking about getting a second call while on line 1, then you just need to enable call waiting on the Asterisk box. The phone should automatically show a second incoming call and allow you to place call 1 on hold. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan Sent: Tuesday, April 12, 2005 7:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 On Tuesday 12 April 2005 10:18 am, MobilPete wrote: can anyone help ?? trying to get Polycom IP300 to utilize both lines, would like calls to roll to open line when incoming call arrives while user is on line 1. Looked everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] QoS TOS numbers and Cisco IOS
tos=0xb8 will set the the packet to be DSCP EF (Cisco likes to use DSCP) Rich Adamson wrote: Does anyone know how setting the TOS bits in iax.conf corresponds to the Cisco TOS types? For example, if I set: tos=0x04 in iax.conf, and on the Cisco, I use: access-list 110 permit ip any any tos 4 I can't get the Cisco to match any packets. I've tried various combinations of numbers on both asterisk and the cisco. I've also tried hex to decimal conversion. I just can't get the Cisco to see the TOS bits that I set in iax.conf. Here's what I'm using. sip.conf: tos=0x18 ;lowdelay ;sets ip tos bits (=lowdelay, throughput) iax.conf: tos=lowdelay Cisco: class-map match-all voice-rtp match access-group 103 access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 C1750#show access-list 103 Extended IP access list 103 permit ip any any tos min-delay (2077271 matches) permit ip any any tos 12 (651833 matches) The NAI Sniffer does a better job of showing the bits. Here's two samples for the above: sip packet (tos=0x18): IP: Type of service = 18 IP: 000. = routine IP: ...1 = low delay IP: 1... = high throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion iax packet (tos=lowdelay): IP: Type of service = 10 IP: 000. = routine IP: ...1 = low delay IP: 0... = normal throughput IP: .0.. = normal reliability IP: ..0. = ECT bit - transport protocol will ignore the CE bit IP: ...0 = CE bit - no congestion Study the above and the bits become very clear. :) ___ 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 -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] binding Asterisk to virtual IP
Xu Wang wrote: Hello Our Asterisk works fine with 'real' IP. But when we change the domain to a virtual IP, the audio stream probably goes to the 'real' IP. There is no sound coming back. Asterisk log shows that it does not hang up. Do you know what might be wrong? Did you look at rtp.conf? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] append # to dial string
John Breeden wrote:
Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how you
can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? -
hex/octal whatever? I'm surprised that I can't find anything
searching the wiki or google.
Try something like this:
exten = _9XXX,1,Dial(Zap/4/${EXTEN}#)
Then you are doing something wrong. The above syntax is correct.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards
[EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. ___ 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] CDR and TDS
David Masure wrote: Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds but when I compile, I've got this message Does anyone have the same problem and/or know how to solve it ? Update of /usr/cvsroot/asterisk/doc In directory mongoose.digium.com:/tmp/cvs-serv24936/doc Added Files: README.tds Log Message: Add documentation for TDS noting compilation problem on 0.63+ --- NEW FILE: README.tds --- PLEASE NOTE The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. The cdr_tds module is known to work with FreeTDS version 0.62.1; it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug fix releases. The cdr_tds module uses the raw libtds API of FreeTDS. It appears that from 0.63 onwards, this is not considered a published API of FreeTDS and is subject to change without notice. Between 0.62.x and 0.63 of FreeTDS, many incompatible changes have been made to the libtds API. For newer versions of FreeTDS, it is recommended that you use the ODBC driver. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Why 's' doesn't take over unknown extension incontext ?
Steve Mann wrote: I think it is i you want, s is the start for a context, meaning anything coming in through that context will start there, i is invalid, and fires if an invalid extension is keyed in that context. s is run when a call comes in and Asterisk does not know the dialed number. It does NOT mean meaning anything coming in through that context will start there ___ 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] timed Loop
Race Vanderdecken wrote:
This might seem really dumb but tack enough silence onto the back of
your file to make it five minutes long. Then the message play for 5
minutes and repeats.
Race The Tyrant Vanderdecken
This was a dumb idea.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Monday, April 11, 2005 11:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] timed Loop
I need to make a time loop in the Extensions.conf. I want it to
play a file every 5 minutes on a call. If I can't use wait because it
ignores all audio. Anyone have any suggestions?
[auto-attendent]
;
; Auto Attendent
;
exten = s,1,SetVar(SAVED_CONTEXT=incoming)
exten = s,2,SetVar(COUNT=1)
exten = s,3,Answer
exten = s,4,DigitTimeout(5)
exten = s,5,ResponseTimeout(7)
exten = s,6,Wait(.5)
exten = s,7,Background(if-u-know-ext-dial)
exten = s,8,Background(company-dir-411)
exten = 411,1,Goto(extensions,2110,1)
exten = 0,1,Playback(pls-wait-connect-call)
exten = 0,2,Goto(extensions,2100,1)
exten = t,1,GotoIf($[${COUNT} = 3]?exit,1)
exten = t,2,SetVar(COUNT=$[${COUNT} + 1])
exten = t,3,Goto(s,7)
exten = i,1,GotoIf($[${COUNT} = 3]?exit,1)
exten = i,2,SetVar(COUNT=$[${COUNT} + 1])
exten = i,3,Playback(extension)
exten = i,4,SayDigits(${INVALID_EXTEN})
exten = i,5,Wait(.5)
exten = i,6,Playback(pbx-invalid)
exten = i,7,Goto(s,6)
exten = exit,1,Playback(goodbye)
exten = exit,2,Wait(3)
exten = exit,3,Hangup
include = extensions
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Re: [Asterisk-Users] Play Sound File Without Answer Channel
Angel Diaz wrote: Mikael, Well, to be more specific, I'm using ISDN PRI. 30B+D. - Original Message - From: Angel Diaz [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 3:55 PM Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel I'm using Zap channels. Does Zap channels support ? Yes, but it would depend on your provider. ___ 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] timed Loop
The ONLY way to MAYBE play an announcement DURING a call is by using the
stuff put in for calling cards. See show application dial
Chris wrote:
That won't work on outgoing calls, will it?
Regard,
Chris
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, April 11, 2005 2:46 PM
Subject: Re: [Asterisk-Users] timed Loop
[auto-attendent]
;
; Auto Attendent
;
exten = s,1,SetVar(SAVED_CONTEXT=incoming)
exten = s,2,SetVar(COUNT=1)
exten = s,3,Answer
exten = s,4,DigitTimeout(5)
exten = s,5,ResponseTimeout(7)
exten = s,6,Wait(.5)
exten = s,7,Background(if-u-know-ext-dial)
exten = s,8,Background(company-dir-411)
exten = 411,1,Goto(extensions,2110,1)
exten = 0,1,Playback(pls-wait-connect-call)
exten = 0,2,Goto(extensions,2100,1)
exten = t,1,GotoIf($[${COUNT} = 3]?exit,1)
exten = t,2,SetVar(COUNT=$[${COUNT} + 1])
exten = t,3,Goto(s,7)
exten = i,1,GotoIf($[${COUNT} = 3]?exit,1)
exten = i,2,SetVar(COUNT=$[${COUNT} + 1])
exten = i,3,Playback(extension)
exten = i,4,SayDigits(${INVALID_EXTEN})
exten = i,5,Wait(.5)
exten = i,6,Playback(pbx-invalid)
exten = i,7,Goto(s,6)
exten = exit,1,Playback(goodbye)
exten = exit,2,Wait(3)
exten = exit,3,Hangup
include = extensions
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Re: [Asterisk-Users] Play Sound File Without Answer Channel
Angel Diaz wrote: I want to use the Voicemail app and before that, I would like to play an audio file but not billable in the Switch side. Than, to do so, I have to be able to no send the Answer message during the play of the file. Then after finish the file, I'w xecute the Voicemail app. That's why I need to play the file before answer the channel. Is it possible ? I have looked at the Playback and Background app, and I see they are answering the channel before playing the file. With analog Zap you cannot do this. With ISDN you can do this (if you provider allows you to). See the noanswer option to Playback. Background should not support this since you can only SEND audio before answer, not receive audio/DTMF and Background is only used when you need to reveive DTMF. ___ 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] Cannot open chan_zap:
Tim Connolly wrote: Well crapola... cvs-head works with Digium's te110xp, but not cvs stable. Looks like there's a huge difference: Stable=-rw--- 1 root root 248572 Jun 9 2004 chan_zap.c Head =-rw--- 1 root root 326585 Apr 6 14:17 chan_zap.c I run a te110p with 1.0.x CVS stable all the time. You have a problem with your modules.conf and forgot to put the .so on the load = line. ___ 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] SIP outgoing problem
snacktime wrote: On Apr 10, 2005 5:28 PM, Paul [EMAIL PROTECTED] wrote: I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick up the handset I get a dialtone, however, when I press 9, the dialtone stops. I assumed it would pause for a moment and give me another dialtone for dialing my outside number. This phone has a softkey that's labeled Dial.if you dial a number, you have to either wait for a few moments for the line to pick up, or press the dial softkey and it will dial the number. If I dial 92125551212 and then press dial, it will dial out. I would like to be able to pick up the handset, press 9, get a dialtone and then dial the 10 digit number without having to press any softkeys. Does anyone have any ideas?? Paul This should help... http://www.voip-info.org/wiki-Asterisk+cmd+DISA Talk about the blind leading the blind. You configure the dialplan for your SIP device ON THE SIP DEVICE. DISA is an ugly hack and should only be used to provide dialtone to devices in the case of the device being too stupid to have a configurable dialplan. The SIPura devices have a powerful dialplan featureset. You just have to read the docs to understand them. SIPura has the docs for their products online. Read them. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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: no ring on inbound SIP calls
Rich Adamson wrote: On incoming SIP calls, the caller just gets silence instead of ringing until * answers the channel. Is this a configuration issue on my end? Chris Correction, this is true for both IAX and SIP incoming calls on my system. I have IAX setup with teliax and SIP with livevoip. Hmm, I did not realize that the Ringing command can be used before the call is answered, I thought it could only be used after it was answered. Putting the Ringing command at the top of the extension fixed my problem. I don't believe an iax - sip call is considered answered until the sip phone picks up. Therefore, the r option instructs asterisk to provide ringback tone until the sip does answer. (That's not necessarily true with some other channels though.) No. r instructs Asterisk to provide a fake ringback tone. If you need r then something is seriously wrong. Asterisk will always provide rinback tones when it thinks it should. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SIP outgoing problem
Tony Hoyle wrote: Eric Wieling wrote: You configure the dialplan for your SIP device ON THE SIP DEVICE. DISA is an ugly hack and should only be used to provide dialtone to devices The OP's question is not answered by modifying the dialplan. He specifically wanted to get a dialtone after dialling 9, not merely to have the sip device send the 9 immediately. DISA is the correct answer. You mean like the , in the SIPura dialplan which says to continue the dialtone? I.e. 9,nxx Do you have a SIPura? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SIP outgoing problem
Paul wrote: I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick up the handset I get a dialtone, however, when I press 9, the dialtone stops. I assumed it would pause for a moment and give me another dialtone for dialing my outside number. This phone has a softkey that's labeled Dial.if you dial a number, you have to either wait for a few moments for the line to pick up, or press the dial softkey and it will dial the number. If I dial 92125551212 and then press dial, it will dial out. I would like to be able to pick up the handset, press 9, get a dialtone and then dial the 10 digit number without having to press any softkeys. Does anyone have any ideas?? You need to look at the dialplan on the SIPura device. The docs for the SPA-841 are not very complete when it comes to dialplans, but SIPura uses the same dialplan syntax across all their products. Here is the (very simple) dialplan on one of my SIPura phones: (9,1[2-9]xx[2-9]xx|9,[2-9]xx|[2-9]xxx) Notice the , That tells the SIPura to continue the dialtone after it gets a leading 9. As you can see, there are no overlapping patterns, so as soon as what you dial matches a patterns the SIPura will dial. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Dialing With Backgound Music
Ugur GUNCER wrote: How can play music when is clients phone ringing in dial progress. Usually you read the documentation. At the Asterisk CLI do a show applications to show you what Asterisk apps are available. Also see musiconhold.conf.sample in the Asterisk source directory (in the configs directory). To see detailed help for a specific application, like Dial, do show application dial. Pay special attention to the m option to Dial. Don't worry about the t option at this time (and don't use that option). The t option is actually a good one. If someone tells you to use it, you can pretty much assume they are a newbie and should take their advice with a grain of salt. t and T are only for a SPECIFIC type of call transfer and most people don't need it. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] s extension doesn't work with ata
Drew Einhorn wrote: The ATA generates it's own dialtone, and waits for the user to dial a number, before sending anything to the * box. So one of the first examples in the in the Brief Introduction to Dialplans from Vol. 1 of the Asterisk Documentation Project. [incoming] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() does not work. The ATA generates a Dialtone and waits for the user to dial, then as soon as the user presses some keys. The ATA sends that extension was not found in [incoming] This example is elaborated into a simple example IVR. But how do we get the intial prompt to play on an ATA? In MY extensions.conf I have a comment above [incoming] that says something like Calls without a destination number land here, usually from the PSTN. s is ONLY EVER called when Asterisk doesn't know what number was dialed. This (generally) only happens if a call is coming in on an ANALOG port, or if the call is coming in on a T-1/E-1 port that does not have DID/DDI service on it. An IP Phone or ATA normally send the number dialed to Asterisk and therefore if you dial 5551212 then the ATA will send the call to exten = 5551212,1,Blah( Now if your ATA is not sending the correct numbers or not waiting for you to finish dialing then the problem is with ATA and NOT Asterisk. You didn't bother to tell us what ATA you are using, so I can't really give you any more advice. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] DS3000P - 20 E1 capacity on single card
izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SPA and NAT traversal
Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SPA and NAT traversal
Jim Sturtevant wrote: Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... nat=yes makes Asterisk use the public IP that is inserted by the far side NAT router instead of the private IP the SIP device puts in the packet. Perhaps there is a problem in your sip.conf that is causing the SPA's packets not to match anything. sip show peers will tell you if Asterisk is seeing the public or the private IP of the far end SPA. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] OT: ManxPower 2005 European Tour
I've helped a lot of people on the mailing lists and on IRC #asterisk. and wanted to let people know that I will be in Europe between May 19 and June 21. Stockholm (VON 2005), Brussels (holiday/vacation), Amsterdam (holiday/vacation), and Madrid (Astricon). There are several weeks during my trip that I have no current plans for and may add other cities to my itinerary. I'm looking for recommendations for lodging and tourist activities in all of the above cities. I would be interested in meeting Asteriskers for drinks or coffee in any of these cities. I am also looking for employment in Europe. I would prefer the Benelux area, but all serious offers will be considered. I have experience in a number of areas including Asterisk/SIP/IAX (2 yrs), Linux (10 yrs), WAN/Frame/T-1/DSL (10 yrs), and more. I can do limited programming in C, Perl and PHP. I am a citizen of the USA and want to relocate to Europe. Eric Wieling [EMAIL PROTECTED] -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Any opinions on quality/service of Teliax?
Brian McSpadden wrote: On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote: I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well? That's the only thing they do that I could do without. But, for the service they provide, I'll gladly pay it. ___ With Teliax I noticed that the delay between the Dial command running and me hearing the ringback tone is unusually long. Not TERRIBLE, just unusual for a VoIP connection. More like the delay when dialing out of an analog port, but they don't use analog ports. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] DS3000P - 20 E1 capacity on single card
Andrew Kohlsmith wrote: On April 9, 2005 02:13 pm, Eric Wieling wrote: izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. I don't think they will; their quad T1/E1/J1 have no such POS on them. Which specific Digium card does not use the TigerJet chip (as shown in lspci)? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] DS3000P - 20 E1 capacity on single card
Andrew Kohlsmith wrote: On April 9, 2005 08:25 pm, Eric Wieling wrote: Which specific Digium card does not use the TigerJet chip (as shown in lspci)? TE405P: 05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) I imagine the TE410 and TE110 are both also similarly lspci'd. I sit corrected. The 4-port T-1/E-1 cards do use the Xilinx. The 1-port cards and 4-port TDM cards do not. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Difference Between NAT=yes and QUALIFY=yes and STUN...
Matt wrote: I have a STUN server running on my Asterisk box which seems to work for most of my SIP clients.. but some of them seem to require NAT=yes turned on. If I go further and turn QUALIFY=yes to on, is there a reason I need to keep running a STUN server? If so, what's the difference? I never understood why Asterisk users seem to have such a fetish for STUN and SER. Most people don't need them. If you have many phones behind NAT and you want the phones to call each other and you want to enable reinvites then, yes, you need SER or STUN or something like that. Asterisk seems to be commonly used in three ways: 1) Home Phone System 2) Business Phone System 3) Internet Telephony Service Provider Generally none of these types of use has a large percentage of phones behind NAT and calling each other. Companies like FWD, etc DO need this since most of their users are calling each other. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Call from publicIP to PrivateIP
Andy Hamilton wrote: I imagine that you are using SIP, which has numerous issures with NAT. Consider using IAX2; one of it's benefits is working with NAT, which I gather is your problem. Or he could just read the Wiki and the mailing list archives to see the simple fixes for a lot of NAT related issues. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] codec translation hints
snacktime wrote: So far it seems that the major thing affecting voice quality on my * box is codec translation. How much cpu is required to translate even a single channel without getting static like sounds or other obvious translation issues? I know this probably depends on the codecs involved, but are there any general guidelines to follow? Unless your machine is slow (under 800Mhz) or you have many calls (more than 2), you are not going to have issues with transcoding. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Warning, flexible rate not heavily tested!
Ronald Wiplinger wrote: Any idea? -- SIP Seeding peers from Astdb: '3366' at [EMAIL PROTECTED]:64440 for 3600 -- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366 -- SIP Seeding peers from Astdb: '886229421761' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.18 for peer 886229421761 Ouch ... error while writing audio data: : Broken pipe Warning, flexible rate not heavily tested! Segmentation fault (core dumped) This is not an Asterisk message. It's a mpg123 message. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Beeps during Sip to Sip phone calls
Daryll Strauss wrote: Yep, I've seen it and from reading http://www.voxilla.com it's a pretty common problem. If you turn on debugging what you'll see is that the Sipura has mistakenly detected a DTMF code in the audio stream and is relaying it by repeating the signal (very loudly I might add) So this appears to be a bug in the most current firmware. I've reported it to Sipura including the debug output. Maybe more people should do the same. You'd think that switching to RFC2833 DTMF would fix that. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Liveviop problem
Andrejus Stavickis wrote:
Hi,
On the iax2 show registry I only see an entry for my SixTel account,
no livevoip.
This is all I received from them on my account activation:
Example for your dial plan:
exten =
_1NXXNXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,2,Hangup
Does not say anything about requirement to put anything else in any
other config file.
I also have a voipjet account and it works fine without any other
entries in any config files other than extensions.conf. Do I in fact
have to register livevoip the same way as SixTel ?
You only need to register to receive calls from your DID (if you have one)
--Eric
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] PRI Advice...
Matt Loretitsch wrote: Looking for some help any way I can. I've been closely following digium's troubleshooting steps and seem to be okay there. I am connecting, via PRI, to a Definity system. When I release the board on the Definity side I get this in Asterisk: *CLI Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395 pri_dchannel: PRI got This appears to be a problem with some specific motherboard. What motherboard (brand AND model) are you using? -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] IAX2 and NATs that increment ports
CuPoTKa wrote: Hello! Does anybody tried to work with IAX2 (client side - softphones) behind a NATs that always increment ports? At asterisk CLI I see: -- Registered '12345' (AUTHENTICATED) at a.b.c.d:22269 -- Registered '12345' (AUTHENTICATED) at a.b.c.d:22289 -- Registered '12345' (AUTHENTICATED) at a.b.c.d:22351 And that clients expects problems, for example if they try to reconnect with softphone - they can't connect any more, and so on. Is there any workaround for such NATs (but we can't touch routers, only asterisk side or something). This is the way NAT works. It's not a problem for Asterisk unless you are doing something silly like port forwarding 4569/UDP on your NAT router. Asterisk doesn't CARE about the source port of the client. ___ 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] Stopping Retransmission Found: 102 Error with Polycom IP300
Min Hwan Chang wrote: Evening, I'm having problems with a Polycom IP300 giving me a Stopping Retransmission Found:102. It gives this error about every 30 seconds. After searching the Help list, I went ahead and set Disallow=all and allow=ulaw. This still doesn't seem to help. Is this problem related to the phones Expiration Time? Or to the millisecond timing of the Ulaw protocol? This error means I'm not getting responses from the device. Usually this is a NAT issue. ___ 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] Dialogic D/300SC-1E1 and D/600SC-2E1 with *
Richard Dutton wrote: I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these particular model and would like to use them in an Asterisk server. YESTERDAY there was a thread that talks about Dialogic cards. Real yesterday's mailing list at http://lists.digium.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] dial out and all circuits are busy
J. Arnaud wrote: Hi, I am using the dial out feature (/var/spool/asterisk/outgoing) but when I look in CDRs, calls that reached a all circuits are busy now, please call later are considered as ANSWERED. Is it the expected behavior? It there a way to change that? If you have analog calls are considered answered as soon as they finish being dialed. ___ 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] how can i connect a cost display on asterisk
Hakem Taourchi wrote: Hello, Do you confirm there is a way to send information and update it while the call is ongoing using the caller Id information ? I strongly doubt this will work on anything except an analog phone. I also strongly doubt that Asterisk supports this at all. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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: busy line status on CISCO 7940/7960
Sergio wrote: Telnetting the phone I see a good amount of free memory space. subscribe/nority is just a firmware implementation. I think it's just a market choice. They wanna sell their new phones with that feature on. What new phones do that have? ___ 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] multiple PBXs on one server.
Scott wrote: Is it possible to run more than one Asterisk PBX on a single server? I don't think there would be a hardware restriction using modern gear but is there limitations on installs etc? I know it would be trivial to make multiple databases for AMP and likely use different ports for the SIP proxy. Anyone accomplish this? They are called contexts and are talked about is practically every single piece of Asterisk documentation that exists. ___ 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 sounds
Josiah Bryan wrote: On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote: Hello all, I am looking for a list of all available sound files for asterisk and a transcription of their content, so that I can have someone translate them into portuguese. I vaguely remeber reading some file in my server that had a list of all the sound files and their transcripts...i just spent about 20 minutes looking for it in the /usr/src/asterisk CVS tree that I checked out - cant seem to find it off hand. Any body have any idea what that file is? It's called sounds.txt ___ 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] Should PRI running over t100p be able to survive short yellow alarms?
Kris Boutilier wrote: I have a PRI connection between Asterisk and a PBX. The connection passes through a hardware echo canceller which includes some monitoring facilities. Occasionally the T1 has gone yellow for short periods (2 seconds) and when this occurs Asterisk seems to immediately tear down any in-progress calls. Is this expected behavior or should the established B-channels stay up (sans audio) until the T1 either goes red or a longer period of time (say 10 seconds) passes? I appreciate the best solution would be to fix the cause of the intermittant yellow alarms, however I am curious... Check your Zaptel card timing setting. i.e. span=1,1,0,esf,b8zs in /etc/zaptel.conf. The second 1 on the line says get timing for T-1 frames FOR THIS CARD from the other end of the T-1 I have a problem where I have two T-1s from our provider and I'm getting the frame timeing from T-1 #1. T-1 #2 every once in a while goes into yellow alarm for a very short time. There's nothing I can do about it because I'm using a 4-port Digium card and there's only one timing source per card. ___ 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] Time sync on PRI
Tobias Jönsson wrote: On Thu, 31 Mar 2005, Peter Svensson wrote: It would not be very hard to add both features to libpri. Libpri already has a function to decode and dump the time/date information. If I remember correctly the time/date IE should be added to the SETUP messages. I have been thinking about adding it, but have not had the time. It's already there, in bristuff patches. Please encourage Digium to add Junghanns' patches to the asterisk code :) Please encourage Junghanns to disclaim the code so it CAN be included in the Asterisk code. ___ 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] ZAP problem (No channel type registered for 'Zap')
Maik Hassel wrote: Hello everybody, I am having trouble setting up a SIP/analog phone gateway. The SIP phones are working, just the Zaptel card doesn't seem to work. I am using the zaptel TDM400P with one FXO module on the last bank (should be channel 4 I suppose). When I try to dial out (either via console or using SIP, I get the following error: --- Asterix console error - *CLI dial 9 -- Executing Dial(OSS/dsp, Zap/4) in new stack Apr 4 10:18:16 WARNING[6294]: channel.c:1901 ast_request: No channel type registered for 'Zap' Apr 4 10:18:16 NOTICE[6294]: app_dial.c:746 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Try using Zap/1 The order of the modules can be confusing. ztcfg -vvv show tell you something useful as well. ___ 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] monmp3thread: Request to schedule in the past?!?!
Glenn Powers wrote: I keep getting this error every five minutes: Apr 4 13:35:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:40:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 4 13:40:02 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! I'm running CVS-v1-0-03/08/05-09:27:38. How can I fix this? Ignore it or upgrade your system to something faster. It's a HARMLESS message. ___ 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] monmp3thread: Request to schedule in the past?!?!
Steve Mann wrote: From what I have read, you made a small mistake, if you are not using Digium hardware, but want to use MeetMe of Music on Hold, you still require a timing source, regardless of kernel. A Zaptel Timer has not been required for MoH for at least a year. ___ 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] rookie getting started question
Randy Paries wrote: Thanks for the info OK my first questions I have edited my zaptel.conf fxsks=1-2 loadzone = us defaultzone=us I have two X100P cards installed When I run /sbin/ztcfg ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? If I set fxsks=1 it works ok. Do an lspci and make sure your BIOS is seeing both cards. You can do an lspci -v to see what IRQs the cards are on. ___ 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] bandwidth
Actually about 80k-82k when you take into account UDP and RTP overhead and assume you are using SIP. Single IAX2 call may be a little less. multiple IAX2 calls using trunking will be a lot less. In fact, this question is answered on http://www.digium.com/index.php?menu=documentation specifically the link to http://www.packetizer.com/voip/diagnostics/bandcalc.html Unfortunatly the above URL is not terribly clear and understandable. People complain about Asterisk's lack of good, organized, understandable documentation. It might help if they actually used the documentation and links that ARE available. Here we have an example of one person that didn't do the research (understandable, since he/she might not have known about the Documentation link on Digium's web site) and then asked a question and then another person that ALSO didn't do the research (I'm guilty of this too, but am getting much better) but answered the question anyway. William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. ___ 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] bandwidth
Bernie wrote: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. Yes. Read up on the various codecs and how much bandwidth they use. ___ 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] Detecting Downed SIP Phone
John Goerzen wrote: Hi, I recently encountered an odd situation: the network cable to my SPA-841 got unplugged while it was in the midst of a call. I got it re-plugged in about 30 seconds, and the phone rebooted. The phone showed no evidence of the previous call in progress and worked like normal. Asterisk, on the other hand, believed the call was still in progress -- my outgoing line was still in use, and it showed up in the show channels list. I resorted to the soft hangup command to terminate it. What could I do so that Asterisk would automatically terminate a call in these situations? Wait. Asterisk will eventually realize the call is gone. That's what the whole Maximum retries exceeced message is about. ___ 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] Transient SIP Registration Issues
Richard J. Sears wrote: Hey Everyone - I am having a problem that is keeping me awake at night.ok, so maybe not keeping me awake, but it is frustrating. :-) I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel 700Mhz box with 512MB of RAM. The system is very light, with maybe 35 SIP and IAX connections. I am using NuFone and Konfer for dialtone with no traditional TDM cards installed at all. Overall system load is around .4 or less most of the time. Overall - a very simple configuration. I am using (mostly) the Linksys PAP2-NA units for deployment. I preconfigure the units, then ship them out to the people that need them. I also have several of the Digium IAXy units in use. The problem I am starting to see is that a person's extension will work great, and then I will start to see failed registrations for their unit over and over again. When this happens, the units fall offline. Then the unit will magically reregister and start to work again. I had assumed (initially) that it was a bad unit, so I replaced it, but then it started to happen to other units as well. When registered, the units in question have ping time under 50 to 60 ms, and no latency associated with them. Packet loss is extremely minimal or none at all. There was a thread recently, I don't recall if it was -users or -dev about SIP registrations failing when network latency was high. I don't remember seeing a solution. I would suggest trying Google, but I'll bet it has not indexed the mailing list archives recently. Try searching manually. --Eric ___ 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 does asterisk know the did called on?
Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your carrier can tell you how many digits they will send to you.
Asterisk sees these digits and will match exten = 1234,1,Blah if the
carrier sends you 4 digits.
Remember Asterisk does not really support DID on analog ports, only
T-1/E-1 (including PRI) ports , BRI ports, and VoIP ports.
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Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote: Hello all, I was hoping to be able to call a mobile and if it is un-reachable for whatever reason (e.g. switched off) then I was expecting an unobtainable response that would be detected in Asterisk. It seems that the operator (Virgin in UK) imedately completes the call and plays an automated message before clearing the call. Does anyone know if there a way of avoiding the call completion for mobiles? I have noticed that Sipgate charge for a calls to an unavailable mobile regardless. Bellsouth at least WILL play an automated message, but NOT answer the line. I work around this by adding the r option to the Dial command. The r option of course provides a fake ringing sound to the caller, even if it REALLY should be doing something else like playing telco audio before answer, or a busy tone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SET CHECK group
Mark Halverson wrote:
exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to
each phone.
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Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
Rod Bacon wrote: This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. That's what I would expect. ___ 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] patlooptest: Usage, setup?
Eric Wieling aka ManxPower wrote: Does anyone know what I need to do to use patlooptest? I have what I think is a T-1 loopback plug in the card (1-port, TE110P), but I still see a red alarm. Is this normal? I don't even know where to start for this. From Digium Support: You will need to specify each span as span=1,0,0,esf,b8zs. You must change the span number of course. Then you will specify clear=1-24 for a T1 or clear=1-31 for an E1. The only other options you should have in your zaptel.conf is loadzone and defaultzone. It does not matter what these are set to. Then you will have to reload the zaptel kernel modules. Run make tests in your zaptel source directory. You will need a T1 loopback cable plugged into the back of the card. Once the T1 loopback cable is installed the span should go green. You may check the status by using zttool. If the span is not green then your T1 loopback cable is faulty. You can make a T1 loopback cable using wires 1 to 4 and 2 to 5. You will run ./patlooptest /dev/zap/1 180. The 180 is the length in seconds that the test will run. /dev/zap/1 is the first clear channel on this span. If you wish to test a second span then you would start with the first clear channel of that span. Span 2 would start at /dev/zap/25 on a T1. patlooptest will only output on errors. Disregard any errors the first 15 seconds of the test. Very few errors over a long period of time are normal. ___ 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: Problem with dial out via chan_capi
Kib Eki wrote: Thanks, problem solved, I found somethind in this mailing list! Wrong extensions.conf entry. extensions.conf: exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr ?? But, what does ,5,tr mean ?? 5 tells Asterisk to hang up if the call is not answered in 5 seconds. t tells Asterisk to use that horrible # hack to do transfers r tells Asterisk to send a ringing sound to the caller, even when doing so is not the right thing to do. show application dial will tell you about the options. ___ 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: Livevoip still no DTMF?
Brian Litzinger wrote: iax.conf: [general] bandwidth=high allow=all jitterbuffer=no tos=low register = 1234567:[EMAIL PROTECTED] [livevoip] type=friend secret=1234567890 deny=0.0.0.0/0.0.0.0 permit=217.160.244.186/255.255.255.0 context=from-livevoip sip.conf: I have dtmfmode=inband for both sip.media.com and sip.broadvoice.com and both are limited to ulaw, alaw. Get rid of the bandwidth= statement. In the [livevoip] put disallow=all and allow=ulaw (or the ONE codec you want to use). Also comment out the tos=low just to see if that makes any difference. ___ 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: Livevoip still no DTMF?
Brian Litzinger wrote: On Fri, Apr 01, 2005 at 12:12:57PM -0600, Eric Wieling aka ManxPower wrote: Brian Litzinger wrote: iax.conf: [general] bandwidth=high allow=all jitterbuffer=no tos=low register = 1234567:[EMAIL PROTECTED] [livevoip] type=friend secret=1234567890 deny=0.0.0.0/0.0.0.0 permit=217.160.244.186/255.255.255.0 context=from-livevoip sip.conf: I have dtmfmode=inband for both sip.media.com and sip.broadvoice.com and both are limited to ulaw, alaw. Get rid of the bandwidth= statement. In the [livevoip] put disallow=all and allow=ulaw (or the ONE codec you want to use). Also comment out the tos=low just to see if that makes any difference. By your command... Made the suggested changes. Called in via SIP and Cell Phone. Still no response to DTMF. It was worth a try. 8-) Try allow=gsm instead, but I doubt it will make any difference. Your other option is to just switch providers. ___ 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: Livevoip still no DTMF?
Brandon Patterson wrote: Level 3 does DTMF inband DTMF. Period. If he's using IAX he's not talking directly to Level 3. ___ 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] Sangoma VS. Digium
Jerry wrote: On Mar 31, 2005, at 8:01 AM, Zoa wrote: cpu load on te4xxp cards is very low, and now that they have echo cancellers as add-ons cards, it will be even lower. I can't speak on hardware compatibility as i never tried a sangoma card. (But i can say that in the last year i've never had an issue with digium cards and we have 8 in use.) The te405p card resolved most incompatibilty issues. Digium has a hardware echo can? Not shipping, according to their online store. ___ 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] Are there online forums instead of this emailforum??
Scott Bussinger wrote: forums). If only we could get people to quit posting in HTML email, life would be grand. :) Mozilla has an option to view ALL messages as text. I use that. I suppose I should not. People that post in HMTL should not get my help. Maybe I can use procmail to send an automated message to anyone that posts a message in HTML. 8-) ___ 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
