Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * --> C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. I gather from your reply that there are recommendations regarding the ethernet connection on your Asterisk server? half-duplex seems bad. Could you elaborate a bit on that? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] abt asterisk
I am working on a project " vide over IP" I am asking you to inform me whether asterisk software PBX supports video over IP hubert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video
Hi! Does * supports video? Especially, SIP or IAX? Is there any cool client for Linux and Windows that is NOT H.323? -- WBR, Max Tulyev (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem - installing TDM400P module
Hello When I tried loading TDM400P module using insmod command, I get following error messages. Is there some problem with my asterisk installation. Please advise. Thanks Tony $insmod wcfxs Using /lib/modules/2.4.20-8/misc/wcfxs.o /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_unregister /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_hooksig /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_transmit /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_receive /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_register
[Asterisk-Users] Asterisk + CRM
Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks -- Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem - installing TDM400P module
On 23 Dec 2003, tony banks wrote: > Hello > > When I tried loading TDM400P module using insmod command, I get > following error messages. Is there some problem with my asterisk > installation. Please advise. Thanks Tony > > $insmod wcfxs > Using /lib/modules/2.4.20-8/misc/wcfxs.o > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_unregister > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_hooksig > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_transmit > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_receive > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_register The wcfxs module depends upon the zaptel module. You are probably better to do: modprobe wcfxs Michael > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem - installing TDM400P module
You could try $ modprobe zaptel $ modprobe wcfxs You need the zaptel bits first. At 09:52 23/12/03, you wrote: $insmod wcfxs Using /lib/modules/2.4.20-8/misc/wcfxs.o /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 trunking on one side only.
I seem to have the same problem now, were you able to resolve this ? joachim. At 22:41 6/11/2003 -0500, you wrote: Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an --LANPSTN will use IAX2 trunking. If I do an "iax2 show trunk" on asterisk1 it says 0 calls on trunk to asterisk 2 (show channels does show the calls). If I do "iax2 show trunk" on asterisk2 it says 7 calls on trunk to asterisk1. I am using GSM and when I look at the traffic using iptraf with 7 calls active from asterisk1 (analog phones TDM400P) to ASTERISK2 Milliwatt() I see asterisk1 transmiting at a little more than 30k above what asterisk2 is transmitting. I have tried peer/friend, notransfer(?),registration/no registration and nothing about the trunking issue changes. Here is my config, some please tell me what I am doing wrong. iax.conf [anistone] type=peer (friend/peer) host=172.16.1.5 (with and without this statement) secret=test2 context=local2 (with and without this statement) trunk=yes extensions.conf exten => 61,1,Dial(IAX2/gateway:[EMAIL PROTECTED]/[EMAIL PROTECTED]) iax.conf [gateway] (I have tried it with this also named anistone) type=peer (friend/peer) host=172.16.1.232 (I have tried it with and without this statement) secret=test context=anistone (with and without this statement) trunk=yes extensions.conf exten => 60,1,Milliwatt() Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem - installing TDM400P module
On Tue, 2003-12-23 at 10:52, tony banks wrote: > Hello > > When I tried loading TDM400P module using insmod command, I get following error > messages. Is there some problem with my asterisk installation. Please advise. Thanks > Tony > > $insmod wcfxs > Using /lib/modules/2.4.20-8/misc/wcfxs.o > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_unregister > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_hooksig > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_transmit > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_receive > /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_register > > That looks like a Red Hat kernel that has a local root exploit iirc so you may wan to upgrade that one. If you haven't done that already, make sure you have the kernel sources installed. Get a fresh copy of zaptel, libpri & asterisk from cvs and then try again. I think the error means that you are trying to load zaptel modules that were build for a different kernel. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Starting Asterisk
Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can either register a speed-dial or register another account. The problem I've found so far, is that speed-dials are not programmed on the phone, but are instead handled by the Call Manager software (not on a user basis, but on a phone, MAC address basis). Likewise, plugging the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red lights (the buttons also light-up red, blue or green), which according to the hidden technical documentation, indicates that the Call Manager is not registering the unit. I can't work out if it's short of firmware embedded in the Call Manager, whether it's searching for a configuration file on the TFTP (Cisco phones need a TFTP to get their settings and SIP firmware), whether it's not happy with the phone being a SIP version, or whether I'm doing something wrong. I've had to learn about the 7960's configuration the hard way, and despite their useless technical documents, have managed to configure most settings. There's quite a bit of extra configuration for the 7960 I'd love to get to, and would like help or advice on. Things like directory services, screen logo, the 7914 and more! If anybody is interested, I have resources and files to; convert from Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail indicator lamp, special key combinations to reset the unit (without pulling the plug out) and locking/unlocking the preferences, configuring the voicemail speed-dial Any help or advice, please let me know! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Starting Asterisk
On Tue, Dec 23, 2003 at 12:18:10PM +, Adthrawn wrote: > Hi, > > Can anybody guide me in configuring the system to start Asterisk from > bootup... Probably a highly remedial question - but you've got to start > somewhere! If you use screen(1), you can do screen -d -m to start asterisk, and able to reattach to to it using screen -d -r. A sample would be like screen -d -m /path/to/asterisk -vgc > > Regards, > Ad. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
I use http://cr.yp.to/daemontools.html. Besides starting asterisk on boot up it keeps an eye on the process and restarts asterisk if it crashes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Tuesday, December 23, 2003 6:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
> >>Packet 50 - Delay 50ms > >>Packet 51 - Delay 5ms > >>Packet 52 - Delay 5ms > >>Packet 53 - Delay 50ms > >>Packet 54 - Delay 5ms > >>Packet 55 - Delay 5ms > >> > >>Is there anyway to space them out evenly at 20ms?? > > > > > > The 20 ms is not the inter-packet timing, its the relative content of what's > > within the packet. In other words, the packet contains 20ms of encoded voice. > > > > If the inter-packet times (delays) are large, as they would seem to be > > in your example, then something else is not right. Possibly a half-duplex > > ethernet connection, something else running on the server, router buffers, > > etc. > > > > On a typical * --> C7960 local call, I generally see from 1ms to 20ms > > inter-packet delays. Seldom (if ever) anything above 20ms. > > > > I gather from your reply that there are recommendations regarding the > ethernet connection on your Asterisk server? half-duplex seems bad. > Could you elaborate a bit on that? Yes, half-duplex ethernet connections "can" cause significant problems depending on the actual load. In very general terms, a half duplex ethernet interface can run up to about 20% utilization before collisions occur, whereas a full duplex connection can run close to 100% without dropping packets. Those rough numbers apply to both 10 meg and 100 meg ethernets. If a collision or dropped packet occurs (in a voip udp environment) there is no way to retransmit the missing/damaged packet. Missing one packet isn't a big deal, but if you have collisions and/or dropped packets, there is a very high probability that lots of packets will be dropped. If too many are dropped, you'll hear the result in the undecoded voice as choppy voice. For whatever reason, most unix systems (and MS systems for that matter) do not give you a convenient way to configure (or even check) "how" your ethernet adapter negotiates the connection. There are no official "standards" as to how the negotiation process determines half vs full, and systems get it wrong about 50% of the time. (As professional network performance consultants, we've diagnosed a very large number of problems like this in corporations around the US over the last ten years. Think in terms of a unix system trying to negotiate half vs full at the exact same time as the switch is doing the same thing "without" actually communicating to the opposite end of the cable.) If the ethernet traffic is low, no one actually notices the problem. But, as traffic increases (multiple RTP sessions, etc) the problem begins to occur and the average technical person doesn't have a clue what is really going on. What makes it difficult to identify/diagnose is that each time the system is rebooted (and each time a Cat 5 cable is disrupted), the half vs full negotiation happens again and (as mentioned) 50% of the time one end gets it wrong. Therefore the performance problem tends to come and go, and support folks typically don't associate the performance issue with the actual half/full problem. (In larger companies, the network support person might reboot a switch without the * support person knowing it, and suddenly the * support person has a problem for which he can't identify what happened.) Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg ethernet would handle roughly 20-25 rtp sessions before bumping into the problem (your milage may vary). The majority of the folks on this list seem to be running home/soho systems and would likely never run into the issue. But the heavier users will. What makes this half/full problem even more difficult to diagnose is that many of these systems have other functions running on them (eg, up2date, remote database calls, web activity, broadcasts) that can consume a fair amount of ethernet bandwidth, and the support person is so highly focused on asterisk they forget some other activity might be impacting their voip quality. Invariably, a Cat 5 cable disruption or reboot (or something else) happens at the same time the support person makes a programming or parameter change, and the person jumps to the conclusion they fixed a problem with their change when in fact the problem was with their ethernet connection. To ensure one never gets bit by the issue, simply ensure that all ethernet interfaces between the asterisk system and the sip phones are statically defined as full-duplex. (Good luck in finding the utilities that let you do that on Linux systems. They are out there, but not easy to find.) The sip phone's negotiation of half vs full is less of an issue as generally the most traffic it sees is one RTP session. But, to obtain maximum smoke and ensure highest quality, the phones should be locked at full duplex as well. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
An even better way to get asterisk started is to use the init scripts provided with asterisk and the zaptel kernel modules. cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel Then do the proper linking, etc to get asterisk to start in your current run level. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David J Carter Sent: Tuesday, December 23, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callwaiting / limits?
> > I'm using grandstream phones, when on a call and a second call comes in > the > > call waiting indication is to play ringing which means you cant actually > hear > > your original call. I want to stop this but cant, heres my options > > > > 1. Change the callwaiting indication, I assume this is produced by the > phone and > > in the case of grandstream there seems to be no way to control this. > > > > 2. Use of incoming/outgoing limit in sip.conf. This works okay except > there is > > no 'absolute limit' type option, meaning that if i place an outbound call > from > > my grandstream it is possible to send a new incoming call in and we have > the > > call waiting again. > > > > I assume others have found this, whats the solution? > > > > Steve > > > > Hi Steve, > > The incominglimit applies to both incoming and outgoing calls, so long as > I'm on the phone, any incoming call gets sent to voicemail. Use the "sip > show inuse" on the CLI to check the inuse counter is being incremented when > on a call, whether receiving or outgoing. > > Is anybody else having this problem ? Ok have looked a bit closer, it works for ordinary calls, my problem is actually with queuing. The call isnt going to the phone (at least the inuse counter stays at 1) so it must either be asterisk adding the ring sound to the stream (doesnt seem likely) or the queue app is ignoring incominglimit i've just started to look at the code to see if i can spot whats going on but this is my first time in doing so for asterisk so i'm not familiar at all with its inner working! :) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tor2 does not load
On Mon, 22 Dec 2003 15:48:37 -0600 Steven Critchfield <[EMAIL PROTECTED]> wrote: > > asterix:~# modprobe tor2 > > Zapata Telephony Interface Registered on major 196 > > Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 > > irq 7 Did not get DONE signal. Short file maybe?? > > Just a guess, but maybe your module file is corrupted. Have you tried > recompiling the module? If that doesn't work, try the standard move > the card to a different slot. Sometimes cards can become belligerent > and will not wake up until they have been initialized in a different > slot. This is not a digium specific trick, but a problem I have had > with other cards. I tried recompiling, but the error is the same. I tried in another machine also. It's strange that lscpi now shows a line that I've never seen before: asterix:~# lspci 00:03.0 Bridge: PLX Technology, Inc.: Unknown device d00d (rev 01) any clue? thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codes/grandstream/PRI.. few questions :)
Hi Guys.. Just wondering if someone could help me with a few questions please. were currently using the ulaw codec with our grandstream/iconnect/asterisk setup and its working pretty good except for the fact it downloads heaps. Does anyone know a good site to get referances to how much each codec downloads/quality etc etc ? Ive tried using that g723 codec but i have have problems as soon as a i dial.. my next question.. :) does anyone know howto fix the grandstream 484 errors you get sometimes when you dial ? i had a look at they rekon to put early dial on.. which just makes things worse heh. They'd be a cool little phone except for this problem. Lastly were looking at getting a PRI or something to handle 30 lines.. I know digium sells hardware to do this, has anyone in australia gotten good results from doing this kind of setup ?? also what are the restrictions in regards to caller id and that sort of stuff in aus? do is all work ? thanks heaps everyone :) Merry Christmas Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
"make config" does both the copy and the neccecary linking... Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: Tuesday, December 23, 2003 8:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk An even better way to get asterisk started is to use the init scripts provided with asterisk and the zaptel kernel modules. cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel Then do the proper linking, etc to get asterisk to start in your current run level. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David J Carter Sent: Tuesday, December 23, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tor2 does not load
Eduardo Goncalves wrote: On Mon, 22 Dec 2003 15:48:37 -0600 Steven Critchfield <[EMAIL PROTECTED]> wrote: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7 Did not get DONE signal. Short file maybe?? Just a guess, but maybe your module file is corrupted. Have you tried recompiling the module? If that doesn't work, try the standard move the card to a different slot. Sometimes cards can become belligerent and will not wake up until they have been initialized in a different slot. This is not a digium specific trick, but a problem I have had with other cards. I tried recompiling, but the error is the same. I tried in another machine also. It's strange that lscpi now shows a line that I've never seen before: asterix:~# lspci 00:03.0 Bridge: PLX Technology, Inc.: Unknown device d00d (rev 01) The dood in question is Jim Dixon, one of the developers of the tormenta 2 card. :-) Thats is your Tormenta 2 card. Regards Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
You should be able to just order Trunk Lines. They are also known as ground start lines. They are usually for incoming only so you would have something like 4-5 Trunk lines for the incoming DID's and the rest would be regular pots lines. In your CAC, you would take the Trunk lines and they would come in on the FXS channels and the POTS lines would come in on the FXO channels. In our area, Trunk lines run about $29-$35 each and then you pay for the DID's. Hope it helps. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Don Pobanz [mailto:[EMAIL PROTECTED] Sent: Monday, December 22, 2003 4:42 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] DID trunks -- equipment requirement On Monday, December 22, 2003 3:40 PM, john lawler [SMTP:[EMAIL PROTECTED] wrote: > Hi guys, > > I posted a somewhat similar question about a month ago and got a > thoughtful resonse from Steven Critchfield, but I've got a quick > follow > up question to it. > > I'm looking to setup a 16 extension / 10-14 phone line Asterisk > install > for a customer who would like to have DID numbers for the extensions, > > since they're currently on Centrex and already have the 1-to-1 > correspondence. Since I'm in a less populated area of the country, > SBC > doesn't seem to have much in the way of fractional T1 products (on the > > scale that we need them) available, Have you asked for a full T1 but with just 10-14 DID/DOD trunks? We can not get fractional T1 here but on a full T1 we can add anywhere from 1 - 24 trunks. So we pay one amount per month for the T1 and on top of that we pay another amount times the number of trunks we have. I know this didn't exactly address your questions. For your primary question I believe that your would need different type of channels in a channel bank than FXOs. DPTs (Dial pulse) terminating come to mind, but that may be wrong. Don Pobanz >so I think my only option for DID > is > to use (analog) DID trunks for incoming calls and POTS lines for > outbound calls. > > I'm familiar w/ POTS lines and have already done limited testing w/ a > CAC channel bank equipped with FXO cards and that works fine. What > I'm > concerned about is the DID trunks. I've been told they have no > dialtone > and of course you can't place calls on them, but can receive calls. > > My question is, in general, should my CAC channel bank w/ the FXO > cards > that work on POTS lines work okay w/ analog DID trunks from the phone > > company? Might I have to purchase additional equipment to handle the > > DIDs (going into one of two Digium T1 cards I have in the Asterisk > box)? Would they be different cards to plug into the CAC channel > bank? > Something totally different? > > Sorry to bring what I know is a rather off-topic question here, but > the > SBC guys don't like to help with customer education so much. As > always, > I appreciate all of your expertise and patience with me and the other > > new guys. > > John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnophone transfer
hello, Is there a way to transfer the call via gnophone, without calling other user and pressing conf on both calls, it seems that all traffic is still going through the gnophone, not that optimal i guess. thanks -- Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
> At present, I have to manually start Asterisk from the command line, > but I'd like to have it automatically start up (and in the correct > mode) at startup. > > Can anybody guide me in configuring the system to start Asterisk from > bootup... Probably a highly remedial question - but you've > got to start > somewhere! I actually use an entry in my /etc/inittab for asterisk. x:2345:respawn:/usr/sbin/asterisk -f -q It starts when I boot, and I can open a console to it with: asterisk -rvvv Another nice advantage to this is that if I want to restart asterisk completely I only have to issue a 'stop now' command and it will respawn after it exits. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold in Conference room?
Hello All, Does anyone here know how I might provide music into a conference room when there is only one participant. Dead silence tends to confuse non-techies who think that they've done something wrong, even after the entry announcement. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 "Lawyers, guns and money can't get me out of this." - Warren Zevon ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sendmail problems
Hello, I'm having some * and sendmail integration problems, probably because i don't know too much about sendmail. My server crashes when I forward voicemail from one * voicemail box to another, everything else works. E-mail notification works on all boxes when new mail arives, the problem only seems to occur during this forwarding function. It's a difficult problem to troubleshoot. If I start * -gc, the server doesn't crash, just hangs up for about 60 seconds then completes the task, so i can't seem to get a core dump to dive into the specifics of what's going on. I'm not sure how to debug sendmail to look at that side. If someone would be kind enough to e-mail me some sample sendmail.cf files, I may be able to see if I'm not configure properly. I've been reading the sendmail.org site but this application is really archain and difficult for me to understand enough to fix it myself. Thanks in advance. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + CRM
We're starting to integrate * with our customer service software. Basically we're pulling off events from the management interface. We're also making some small patches to the code to deliver more events about the channel variables, etc. Anton Yurchenko wrote: Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
Interesting. For the record, the MultiTech MVP-130 comes with a default setting of 60ms packets on all of its supported codecs. I changed the packet sizes to 20ms because I had never heard of anyone using such large sample sizes. Andres wrote: On Monday 22 December 2003 19:58, Rich Adamson wrote: On Monday 22 December 2003 16:37, Andres wrote: On Monday 22 December 2003 15:36, Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * --> C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything. Thanks! Andres Ok...after some more testing, the traffic shaping software was not the culprit. It turns out that if the UA is configured for 60ms of voice, then Asterisk will show this strange behaviour. If we set the UA for 20ms, then all works well. Cool! How did it get set to 60ms? The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the transmit packet size to 60ms (or multiple other values). Asterisk will receive 60ms and transmit 20ms times 3 packets, andit works quite well. In any case our SPA2000 problem was unrelated to the packet time. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
Olle, Here is an interesting site that goes into some of the troubleshooting techniques in Voip: http://www.voiptroubleshooter.com/ Maybe it will help your FAQ! Olle E. Johansson wrote: Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * --> C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. I gather from your reply that there are recommendations regarding the ethernet connection on your Asterisk server? half-duplex seems bad. Could you elaborate a bit on that? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Tue, 23 Dec 2003, Rich Adamson wrote: > If a collision or dropped packet occurs (in a voip udp environment) there > is no way to retransmit the missing/damaged packet. Missing one packet isn't > a big deal, but if you have collisions and/or dropped packets, there is a > very high probability that lots of packets will be dropped. If too many > are dropped, you'll hear the result in the undecoded voice as choppy > voice. Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 hardware automatically resends packets involved in a collision - layer 3 is never aware of it happening (although it may cause additional delay). Eventually the ethernet card will give up if too many collisions occur during retries, but this is very rare in practice unless the network is *VERY* loaded. > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg > ethernet would handle roughly 20-25 rtp sessions before bumping into the > problem (your milage may vary). The majority of the folks on this list > seem to be running home/soho systems and would likely never run into the > issue. But the heavier users will. For a duplex mismatch, my experience is that if one end on a 100 Mb/sec link is half and the other is full, bandwidth is limited to about 8 Mb/sec max. This is based on some tests I've accidentally conducted. If you try to send 9 Mb/sec over that link, yes, some packets will get dropped as they simply won't fit. (But I do agree that for a half-half link, you can get about 20 Mb/sec) -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + CRM
Which events do you refer? Regards, Gus - Original Message - From: "Jonathan Tew" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, December 23, 2003 12:25 PM Subject: Re: [Asterisk-Users] Asterisk + CRM > We're starting to integrate * with our customer service software. > Basically we're pulling off events from the management interface. We're > also making some small patches to the code to deliver more events about > the channel variables, etc. > > Anton Yurchenko wrote: > > > Hello, > > > > Anyone aware of any CRM products projects that intagrete with *? Or > > that integrate with any telephony products? Is there some open API for > > such integration, or are they all proprietory? > > > > Thanks > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PBX Functionality How-to
Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Capi Dial & outgoing msn?
Hi all,
I am trying to get Capi Dial to use a specific outgoing msn. I can't get
it to work. If I make a test call to 0703241494 (same isdn line, just
one of the other numbers) I don't get CLID at all. Any ideas?
; use 0703241432 as outgoing msn
exten => _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r)
in capi.conf I have:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;rxgain=0.0
;txgain=0.0
[interfaces]
msn=0703241432
incomingmsn=703241432
controller=1,2
softdtmf=1
accountcode=
context=default
;echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
devices=2
msn=0703241434
incomingmsn=703241434
controller=1,2
softdtmf=1
accountcode=
context=default
;echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
devices=2
Thanks,
Patrick
___
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RE: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
The problem occurs when the software is expecting the packet in a certain timeframe so that it can reassemble it in a timely manner. It's not a big deal with a web page or something along that lines. But when a voice application cannot get reassembled in a timely manner, you'll surely notice it! -Original Message- From: Joel Maslak To: [EMAIL PROTECTED] Sent: 12/23/2003 10:41 AM Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) On Tue, 23 Dec 2003, Rich Adamson wrote: > If a collision or dropped packet occurs (in a voip udp environment) there > is no way to retransmit the missing/damaged packet. Missing one packet isn't > a big deal, but if you have collisions and/or dropped packets, there is a > very high probability that lots of packets will be dropped. If too many > are dropped, you'll hear the result in the undecoded voice as choppy > voice. Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 hardware automatically resends packets involved in a collision - layer 3 is never aware of it happening (although it may cause additional delay). Eventually the ethernet card will give up if too many collisions occur during retries, but this is very rare in practice unless the network is *VERY* loaded. > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg > ethernet would handle roughly 20-25 rtp sessions before bumping into the > problem (your milage may vary). The majority of the folks on this list > seem to be running home/soho systems and would likely never run into the > issue. But the heavier users will. For a duplex mismatch, my experience is that if one end on a 100 Mb/sec link is half and the other is full, bandwidth is limited to about 8 Mb/sec max. This is based on some tests I've accidentally conducted. If you try to send 9 Mb/sec over that link, yes, some packets will get dropped as they simply won't fit. (But I do agree that for a half-half link, you can get about 20 Mb/sec) -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] abt asterisk
On Tuesday 23 December 2003 02:27, Hubert Kiyimba wrote: > I am working on a project " vide over IP" > > I am asking you to inform me whether asterisk software PBX supports > video over IP IAX explicitly supports images, video, and URLs. See the gnophone client. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Capi Dial & outgoing msn?
Hi,
try it without prefix (else dtag uses first msn) -
so if your city code is 07032 and phone no (msn) 41432
-> exten => _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
Thomas
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von Patrick
> Gesendet: Dienstag, 23. Dezember 2003 16:53
> An: [EMAIL PROTECTED]
> Betreff: [Asterisk-Users] Capi Dial & outgoing msn?
>
>
> Hi all,
>
> I am trying to get Capi Dial to use a specific outgoing msn.
> I can't get
> it to work. If I make a test call to 0703241494 (same isdn line, just
> one of the other numbers) I don't get CLID at all. Any ideas?
>
> ; use 0703241432 as outgoing msn
> exten => _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r)
>
> in capi.conf I have:
>
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;rxgain=0.0
> ;txgain=0.0
>
> [interfaces]
> msn=0703241432
> incomingmsn=703241432
> controller=1,2
> softdtmf=1
> accountcode=
> context=default
> ;echosquelch=1
> echocancel=yes
> echotail=64
> ;deflect=12345678
> devices=2
>
> msn=0703241434
> incomingmsn=703241434
> controller=1,2
> softdtmf=1
> accountcode=
> context=default
> ;echosquelch=1
> echocancel=yes
> echotail=64
> ;deflect=12345678
> devices=2
>
> Thanks,
> Patrick
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
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[Asterisk-Users] perl database get
Does anyone have any example perl agi script that does a database get. I am being thick and can't seem to get the return value: print "DATABASE PUT big bigger biggest \n"; This bit works fine print "DATABASE GET big bigger \n"; Now what do I do to get the my value from the database get?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PBX Functionality How-to
On Tue, 2003-12-23 at 09:48, Christopher J. Wolff wrote: > Hello, > > I had a partner of mine present a Centrex 21 brochure and ask how many of > those features can I fulfill. There is nothing out of the ordinary, it's > stuff like call hold, call forward, 3-way calling, etc. Has anyone > assembled a how-to that shows how to configure PBX or Centrex type > functionality? I found one in the voip-info wiki but only a couple of > topics were filled out. Could you at least read the documentation around here before you ask for someone to do your work for you. If you can't be bothered to read the documentation, at least offer to pay one of the fine consultants on the list to do your work. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
I'm not sure under what circumstances (from an overall performance perspective) 20ms is better then 60ms, or the reverse. Gut feeling would suggest choosing the size is roughly equivalent to MTU size. The 60ms setting should result in larger packets which might be okay for high speed uncongested links and satellite links. However, the smaller 20ms packets effectively allow "more opportunity" for others to talk on the wire and would likely improve response time for all devices on the wire. Rich > Interesting. For the record, the MultiTech MVP-130 comes with a default > setting > of 60ms packets on all of its supported codecs. I changed the packet > sizes to > 20ms because I had never heard of anyone using such large sample sizes. > > Andres wrote: > > >On Monday 22 December 2003 19:58, Rich Adamson wrote: > > > > > >>>On Monday 22 December 2003 16:37, Andres wrote: > >>> > >>> > On Monday 22 December 2003 15:36, Rich Adamson wrote: > > > >>I have a question regarding the Asterisk Packet Time for SIP Calls. > >> It is hardcoded at 20ms but when I do an RTP Analysis on a stream > >>it is clear that these packets are not spaced out at 20ms. In > >>general you see something like: > >> > >>Packet 50 - Delay 50ms > >>Packet 51 - Delay 5ms > >>Packet 52 - Delay 5ms > >>Packet 53 - Delay 50ms > >>Packet 54 - Delay 5ms > >>Packet 55 - Delay 5ms > >> > >>Is there anyway to space them out evenly at 20ms?? > >> > >> > >The 20 ms is not the inter-packet timing, its the relative content of > >what's within the packet. In other words, the packet contains 20ms of > >encoded voice. > > > >If the inter-packet times (delays) are large, as they would seem to > >be in your example, then something else is not right. Possibly a > >half-duplex ethernet connection, something else running on the > >server, router buffers, etc. > > > >On a typical * --> C7960 local call, I generally see from 1ms to 20ms > >inter-packet delays. Seldom (if ever) anything above 20ms. > > > > > Thanks for your Input Rich. I went ahead and tested this on our > production servers and sure enough the inter-packet times are 20ms. > There must be something happening with our LAB Asterisk. It could be > the CBQ traffic shaping software we have running on it. I will fiddle > around with it to see if it changes anything. > > Thanks! > Andres > > > >>>Ok...after some more testing, the traffic shaping software was not the > >>>culprit. It turns out that if the UA is configured for 60ms of voice, > >>>then Asterisk will show this strange behaviour. If we set the UA for > >>>20ms, then all works well. > >>> > >>> > >>Cool! > >> > >>How did it get set to 60ms? > >> > >> > >The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the > >transmit packet size to 60ms (or multiple other values). Asterisk will > >receive 60ms and transmit 20ms times 3 packets, andit works quite well. In > >any case our SPA2000 problem was unrelated to the packet time. > > > >Regards, > >Andres > > > > > >> > >>___ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk , Video Switching
Dear members, I am writing to inquire whether Asterisk can serve as video switching software for the purposes of video conferencing over IP on a campus network. Hubert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
There's no reassembly with udp, and there is no sense of packets arriving in the same order as what was sent. Udp is a best-effort low-overhead way of transmitting data (with UDP often times referred to as the Unreliable Data Protocol). Changing to TCP would allow reassembly, however the overhead would be substantial. > The problem occurs when the software is expecting the packet in a certain > timeframe so that it can reassemble it in a timely manner. It's not a big > deal with a web page or something along that lines. But when a voice > application cannot get reassembled in a timely manner, you'll surely notice > it! > > -Original Message- > From: Joel Maslak > To: [EMAIL PROTECTED] > Sent: 12/23/2003 10:41 AM > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) > > On Tue, 23 Dec 2003, Rich Adamson wrote: > > > If a collision or dropped packet occurs (in a voip udp environment) > there > > is no way to retransmit the missing/damaged packet. Missing one packet > isn't > > a big deal, but if you have collisions and/or dropped packets, there > is a > > very high probability that lots of packets will be dropped. If too > many > > are dropped, you'll hear the result in the undecoded voice as choppy > > voice. > > Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 > hardware automatically resends packets involved in a collision - layer 3 > is never aware of it happening (although it may cause additional delay). > Eventually the ethernet card will give up if too many collisions occur > during retries, but this is very rare in practice unless the network is > *VERY* loaded. > > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg > > ethernet would handle roughly 20-25 rtp sessions before bumping into > the > > problem (your milage may vary). The majority of the folks on this list > > seem to be running home/soho systems and would likely never run into > the > > issue. But the heavier users will. > > For a duplex mismatch, my experience is that if one end on a 100 Mb/sec > link is half and the other is full, bandwidth is limited to about 8 > Mb/sec > max. This is based on some tests I've accidentally conducted. If you > try > to send 9 Mb/sec over that link, yes, some packets will get dropped as > they simply won't fit. (But I do agree that for a half-half link, you > can > get about 20 Mb/sec) > > -- > Joel > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Tuesday 23 December 2003 10:59, Rich Adamson wrote: > I'm not sure under what circumstances (from an overall performance > perspective) 20ms is better then 60ms, or the reverse. Gut feeling would In our network we set UAs to use 60ms (using G729). Actual data measurements indicate a call consumes about 13Kbps. If we use 20ms then it consumes about 25Kbps. These are of course peer-peer calls since Asterisk itself does not support transmitting at 60ms. We prefer 60ms due to the fact that some of our customers are using dial-up for their VoIP, and bigger delays are preferable to dopped packets. Andres. > suggest choosing the size is roughly equivalent to MTU size. The 60ms > setting should result in larger packets which might be okay for high > speed uncongested links and satellite links. However, the smaller 20ms > packets effectively allow "more opportunity" for others to talk on the > wire and would likely improve response time for all devices on the wire. > > Rich > > > > Interesting. For the record, the MultiTech MVP-130 comes with a default > > setting > > of 60ms packets on all of its supported codecs. I changed the packet > > sizes to > > 20ms because I had never heard of anyone using such large sample sizes. > > > > Andres wrote: > > >On Monday 22 December 2003 19:58, Rich Adamson wrote: > > >>>On Monday 22 December 2003 16:37, Andres wrote: > > On Monday 22 December 2003 15:36, Rich Adamson wrote: > > >>I have a question regarding the Asterisk Packet Time for SIP Calls. > > >> It is hardcoded at 20ms but when I do an RTP Analysis on a stream > > >>it is clear that these packets are not spaced out at 20ms. In > > >>general you see something like: > > >> > > >>Packet 50 - Delay 50ms > > >>Packet 51 - Delay 5ms > > >>Packet 52 - Delay 5ms > > >>Packet 53 - Delay 50ms > > >>Packet 54 - Delay 5ms > > >>Packet 55 - Delay 5ms > > >> > > >>Is there anyway to space them out evenly at 20ms?? > > > > > >The 20 ms is not the inter-packet timing, its the relative content > > > of what's within the packet. In other words, the packet contains > > > 20ms of encoded voice. > > > > > >If the inter-packet times (delays) are large, as they would seem to > > >be in your example, then something else is not right. Possibly a > > >half-duplex ethernet connection, something else running on the > > >server, router buffers, etc. > > > > > >On a typical * --> C7960 local call, I generally see from 1ms to > > > 20ms inter-packet delays. Seldom (if ever) anything above 20ms. > > > > Thanks for your Input Rich. I went ahead and tested this on our > > production servers and sure enough the inter-packet times are 20ms. > > There must be something happening with our LAB Asterisk. It could be > > the CBQ traffic shaping software we have running on it. I will > > fiddle around with it to see if it changes anything. > > > > Thanks! > > Andres > > >>> > > >>>Ok...after some more testing, the traffic shaping software was not the > > >>>culprit. It turns out that if the UA is configured for 60ms of voice, > > >>>then Asterisk will show this strange behaviour. If we set the UA for > > >>>20ms, then all works well. > > >> > > >>Cool! > > >> > > >>How did it get set to 60ms? > > > > > >The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set > > > the transmit packet size to 60ms (or multiple other values). Asterisk > > > will receive 60ms and transmit 20ms times 3 packets, andit works quite > > > well. In any case our SPA2000 problem was unrelated to the packet > > > time. > > > > > >Regards, > > >Andres > > > > > >>___ > > >>Asterisk-Users mailing list > > >>[EMAIL PROTECTED] > > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > >___ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ---End of Original Message- > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] turning off IAX registration attempts
I have, in iax.conf the register statement: register => username:[EMAIL PROTECTED] This causes registration attempts to iaxtel.com for both IAX and IAX2. Every once in a while there is a packet for port 4569 keeping the IAX2 registration alive. This is fine. But, I have a barrage of registration attempts to iaxtel on port 5036 for IAX. Every UDP packet is answered with an ICMP packet claiming "port unreachable". I know that iaxtel has turned off IAX, So, how do I turn off the registration attempts for IAX, for that particular connection? (and keep IAX2) Just seems like alot of wasted bandwidth, contiously knocking on a locked door. Ok, not alot of bandwidth, but, completely useless. Has anyone done a tcpdump at iaxtel to see how many IAX registration attempts hit them, and how fast? Here is my tcpdump: there are ICMP return packets for each of these UDP packets [EMAIL PROTECTED]:/etc/asterisk# tcpdump -n ip host 69.73.19.178 and udp port 5036 tcpdump: listening on eth0 17:10:01.740865 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.740912 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.760869 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:01.760909 198.144.196.118.5036 > 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:09.740652 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.201240 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750502 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750535 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.750546 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:11.770504 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:12.220512 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:25.240316 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:26.250264 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:29.740007 198.144.196.118.5036 > 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:31.759849 198.144.196.118.5036 > 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 17:10:39.279658 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:39.749612 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:40.299550 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:41.759498 198.144.196.118.5036 > 69.73.19.178.5036: udp 12 (DF) [tos 0x10] 17:10:41.759546 198.144.196.118.5036 > 69.73.19.178.5036: udp 42 (DF) [tos 0x10] 20 packets received by filter 0 packets dropped by kernel -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Authentication
You have not covered very much of the
configuration that can be done here. So with that I have come up with a
very generic config for you that I have not tested and is to the best of my
memory but I will give it to you as a starting point. I am posting the
extensions.conf, zapata.conf and voicemail.conf.
It may make sense it may not. I hope
it at least helps and does not hinder.
Assuming FXO are channels 1 and
2
Assuming FXS are channels 3 through
5
Since you do not have a direct mapping
between users and extensions I gave users 1-3 direct access to Zap/3-5 and User
4 gets stuck with a voicemail only extension.
You did not mention if you wanted a menu
system for incoming calls so I did not create one. Instead all incoming
calls from either line will just ring all three extensions. If no one
picks up it goes to a generic voicemail box of 1000.
User 1 can dial 9 1234 ??? etc with
1234 being the password for user 1
User 2 can dial 9 2345 ??? etc
with 1234 being the password for user 2
etc...
Now in reality it would probably be a
cleaner and nicer config using the Authentication app that is available to you
but you asked for the user to be able to just dial 9 1-4 phone number. I
chose 4 digit passwords. If you modify that make sure you modify the
${EXTEN:5} to what is needed. the :5 is trimming off the 9 and 4 additional
digits for the password so if you were using 2 digit passwords you would want to
change that to a :3.
Voicemail for each user is mute at this
point as you have no menu system to direct a caller to a specific user hence
voicemail here will be interoffice only at this point until you create a menu
system or direct incoming lines to a specific user.
Use this at your own risk. I did not
try this configuration on any box. I did this from memory and copying and
tweaking some of my configs and my memory basically sucks so take that as you
will. Most of what I know came from samples around the net so you will see
a lot of stuff from various people around the internet. I am a newbie at
this as well and did not see anyone replying to your message so I thought I
would give it a shot at least to get you going in the right
direction.
I am sure I forgot a lot of stuff that you
would need but hopefully I covered what you asked for at least.
* NO FLAMES * NO FLAMES * NO FLAMES * NO
FLAMES * NO FLAMES * NO FLAMES * NO FLAMES * NO FLAMES * NO FLAMES *
I know I use a lot of whitespace and have
been told numerous times not too but my system works as it is supposed
to
so I guess I have the whitespaces in the
proper area. Too bad if it uses more bandwidth here it makes it easier
for
my brain to understand so you will just
have to live with it all. If you don't like the whitespace then don't read
the
email.
Good luck and happy holidays,
Robert
___
;zapata.conf; Channels definitions for
zapata.conf file[channels]
language
= en
; FXO
Channelssignalling
= fxs_ks ; Assuming you are using KewlStart if not change this to what you
use.group
=
1callgroup
=
1pickupgroup
=
1usecallerid
=
yescallerid
=
asreceivedhidecallerid
=
nocallwaiting
= yescallwaitingcallerid =
yesthreewaycalling =
yestransfer
= yescancallforward =
yesechocancel
= yesechocancelwhenbridged =
yesrxgain
=
0.6txgain
=
0.6immediate
=
nobusydetect
=
nocallprogress
=
nomusiconhold
= random; Analog phone line attached to:
???-???-context
=
fxo-line1-inmailbox
= 1000 ; Not mapped to a specific user have them both go to generic vm
1000channel
=> 1 ; X101P Card; Analog phone line attached to:
???-???-context
=
fxo-line2-inmailbox
= 1000 ; Not mapped to a specific user have them both go to generic vm
1000channel
=> 2 ; X101P Card
; FXS
Channelssignalling
= fxo_ks ; Assuming you are using KewlStart if not change this to what you
use.group
=
2callgroup
=
2pickupgroup
=
2callwaiting
= yescallwaitingcallerid =
yesthreewaycalling =
yestransfer
= yescancallforward =
yesechocancel
= yesechocancelwhenbridged =
yesmailbox
=
callerid
= "Zap 3"
<1>context
=
fxo-outchannel
=> 3 ; TDM30B Port
1mailbox
=
callerid
= "Zap 4"
<2>context
=
fxo-outchannel
=> 4 ; TDM30B Port
2mailbox
Re: [Asterisk-Users] sendmail problems
You say "The server crashes" I assume you mean that Asterisk core dumps and sendmail continues to run just fine. If you can send mail out of the box sendmail is confgured well enough and I doubt the problem is there. If you can get Asterisk to dump then what you need to do is use a debugger to get a backtrace. This will tell to the line (as i line of coe) that caused the crash. The thing to remember to that if a program crashed it is due to t bug.. There _should_ be no way for a user through misconfiguration to cause a core dump. What you are looking for is a little bit od C cde that doesn't handle some condition well. If yu use "gdb" and the "bt" commad you can find the line Asterisk was executing when it crashed. I'd not suspect sendmail --- [EMAIL PROTECTED] wrote: > Hello, > > I'm having some * and sendmail integration problems, probably because > i don't know too much about sendmail. My server crashes when I > forward voicemail from one * voicemail box to another, everything > else works. E-mail notification works on all boxes when new mail > arives, the problem only seems to occur during this forwarding > function. It's a difficult problem to troubleshoot. If I start * > -gc, the server doesn't crash, just hangs up for about 60 seconds > then completes the task, so i can't seem to get a core dump to dive > into the specifics of what's going on. I'm not sure how to debug > sendmail to look at that side. If someone would be kind enough to > e-mail me some sample sendmail.cf files, I may be able to see if I'm > not configure properly. I've been reading the sendmail.org site but > this application is really archain and difficult for me to understand > enough to fix it myself. Thanks in advance. > > JR > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk , Video Switching
On Tue, 23 Dec 2003, Hubert Kiyimba waxed: > Dear members, > > I am writing to inquire whether Asterisk can serve as video switching > software for the purposes of video conferencing over IP on a campus network. > > Hubert http://www.gnophone.com/ -- Chris Maj Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key fingerprint = 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PBX Functionality How-to
One thing Centrex is that Asterisk is not is a turn key system. With Asterisk you have to either build the PBX your self or pay someone to build yu one. With Centrex you simply write a check. THat said, you can build anything you want so of cource the feature list can match. The best way to learn what it can do is to build a small PBX with just a couple extensions. Try to build in all the funtionality you need in your larger system. If you get into trouble you may want to ask __specific__ questions like "I want to make XXX work, I triedd XXX and YYY but I still have this problem it it? You may have to post 50 questions like that one at a time. But you will get answers. Asterisk has a learning curve. expect it to take a few weeks of study But the bottom line is that Asterisk will do quite a bit more than Centrex. I don't think Centrex does VOIP at all --- "Christopher J. Wolff" <[EMAIL PROTECTED]> wrote: > Hello, > > I had a partner of mine present a Centrex 21 brochure and ask how > many of > those features can I fulfill. There is nothing out of the ordinary, > it's > stuff like call hold, call forward, 3-way calling, etc. Has anyone > assembled a how-to that shows how to configure PBX or Centrex type > functionality? I found one in the voip-info wiki but only a couple > of > topics were filled out. > > Regards, > Christopher J. Wolff, VP CIO > Broadband Laboratories, Inc. > http://www.bblabs.com > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Tuesday 23 December 2003 11:40, Rich Adamson wrote: > There's no reassembly with udp, and there is no sense of packets arriving > in the same order as what was sent. Udp is a best-effort low-overhead way Right, UDP itself does not care about order, but at the application layer you can keep track of it. You can design your application to buffer X packets and then reorder them according to sequence numbers. > of transmitting data (with UDP often times referred to as the Unreliable > Data Protocol). Changing to TCP would allow reassembly, however the > overhead would be substantial. > > > > > The problem occurs when the software is expecting the packet in a certain > > timeframe so that it can reassemble it in a timely manner. It's not a > > big deal with a web page or something along that lines. But when a voice > > application cannot get reassembled in a timely manner, you'll surely > > notice it! > > > > -Original Message- > > From: Joel Maslak > > To: [EMAIL PROTECTED] > > Sent: 12/23/2003 10:41 AM > > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) > > > > On Tue, 23 Dec 2003, Rich Adamson wrote: > > > If a collision or dropped packet occurs (in a voip udp environment) > > > > there > > > > > is no way to retransmit the missing/damaged packet. Missing one packet > > > > isn't > > > > > a big deal, but if you have collisions and/or dropped packets, there > > > > is a > > > > > very high probability that lots of packets will be dropped. If too > > > > many > > > > > are dropped, you'll hear the result in the undecoded voice as choppy > > > voice. > > > > Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 > > hardware automatically resends packets involved in a collision - layer 3 > > is never aware of it happening (although it may cause additional delay). > > Eventually the ethernet card will give up if too many collisions occur > > during retries, but this is very rare in practice unless the network is > > *VERY* loaded. > > > > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg > > > ethernet would handle roughly 20-25 rtp sessions before bumping into > > > > the > > > > > problem (your milage may vary). The majority of the folks on this list > > > seem to be running home/soho systems and would likely never run into > > > > the > > > > > issue. But the heavier users will. > > > > For a duplex mismatch, my experience is that if one end on a 100 Mb/sec > > link is half and the other is full, bandwidth is limited to about 8 > > Mb/sec > > max. This is based on some tests I've accidentally conducted. If you > > try > > to send 9 Mb/sec over that link, yes, some packets will get dropped as > > they simply won't fit. (But I do agree that for a half-half link, you > > can > > get about 20 Mb/sec) > > > > -- > > Joel > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ---End of Original Message- > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WEBMIN module for Asterisk
Hello, has anyone come across a module for WEBMIN to configure * ? webmin info http://www.webmin.com/ Thanks Doug -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 1003 http://www.pulver.com/fwd/ ext. 83740 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Starting Asterisk
Look in the directory /etc/init.d (/etc/rc.d/init.d on some systems) You put a script in there called "asterisk". There is a sample called "asterisk.init" in the source. copy it to /etc/init.d/asterisk You may want to study the other files in /etc/init.d to see how they work. Next read the "chkconfig" man page and see way you'd want to type "chkconfig --add asterisk; chkconfig asterisl on" Finally to start asterisk you can type "./asterisk start" You may also want to re-boot the computer to verify that asterisk does start automatically --- [EMAIL PROTECTED] wrote: > On Tue, Dec 23, 2003 at 12:18:10PM +, Adthrawn wrote: > > Hi, > > > > Can anybody guide me in configuring the system to start Asterisk > from > > bootup... Probably a highly remedial question - but you've got to > start > > somewhere! > > If you use screen(1), you can do screen -d -m to start asterisk, and > able to > reattach to to it using screen -d -r. > > A sample would be like > > screen -d -m /path/to/asterisk -vgc > > > > > Regards, > > Ad. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: SIP vs. Skinny protocol
I assume there are several people on this list that have Cisco Call Manager implementations under their belt We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two obvious benefits to using SIP: 1. I can get cheaper phones that run SIP, altough Cisco just came out with a 7902G for $130 US. 2. It's an open protocol and is more likely to survive long-term. What functionality do I lose by going with Skinny? Will Cisco eventually go with SIP only and I'll have to convert anyway? Any other pluses or minuses? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Capi Dial & outgoing msn?
On Tue, 2003-12-23 at 17:13, [EMAIL PROTECTED] wrote:
> Hi,
>
> try it without prefix (else dtag uses first msn) -
> so if your city code is 07032 and phone no (msn) 41432
> -> exten => _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
>
>
> Thomas
Thanks for the pointer Thomas. I removed the areacode from msn= in
capi.conf and from the dial statement. Tried again and till no CLID.
Stumped at this point. Perhaps my telco doesn't allow setting outgoing
msn numbers.
Regards,
Patrick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
The reason you use UDP over TCP for realtime meadia is that TCP's ability to reliably deliver every packet in order actually sounds worse. Reason being is that with a UDP system a dropped packet sounds like just a dropout but if you used TCP the audio stream would be held up and delayed in a queue while that lost packet was being retransmitted. In stead of a dropout the audio would sound as if someone kept hitts a "pause" button on a tape recorder. A dropout sounds better then a delay of potentialy several seconds Almost all realtime meadia systems (telephony, video, possition reporting and so on) maintain some kind of a buffer on the recieving end. But you trad the buffer lenght for delay. Using UDP allows the application to do the buffering where as TCP putting this buffing functin in the operaing systems network code. --- Andres <[EMAIL PROTECTED]> wrote: > On Tuesday 23 December 2003 11:40, Rich Adamson wrote: > > There's no reassembly with udp, and there is no sense of packets > arriving > > in the same order as what was sent. Udp is a best-effort > low-overhead way > Right, UDP itself does not care about order, but at the application > layer you > can keep track of it. You can design your application to buffer X > packets > and then reorder them according to sequence numbers. > > > of transmitting data (with UDP often times referred to as the > Unreliable > > Data Protocol). Changing to TCP would allow reassembly, however the > > overhead would be substantial. > > > > > > > > > The problem occurs when the software is expecting the packet in a > certain > > > timeframe so that it can reassemble it in a timely manner. It's > not a > > > big deal with a web page or something along that lines. But when > a voice > > > application cannot get reassembled in a timely manner, you'll > surely > > > notice it! > > > > > > -Original Message- > > > From: Joel Maslak > > > To: [EMAIL PROTECTED] > > > Sent: 12/23/2003 10:41 AM > > > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) > > > > > > On Tue, 23 Dec 2003, Rich Adamson wrote: > > > > If a collision or dropped packet occurs (in a voip udp > environment) > > > > > > there > > > > > > > is no way to retransmit the missing/damaged packet. Missing one > packet > > > > > > isn't > > > > > > > a big deal, but if you have collisions and/or dropped packets, > there > > > > > > is a > > > > > > > very high probability that lots of packets will be dropped. If > too > > > > > > many > > > > > > > are dropped, you'll hear the result in the undecoded voice as > choppy > > > > voice. > > > > > > Actually, collisions occur at Layer 2, not Layer 3, and the layer > 2 > > > hardware automatically resends packets involved in a collision - > layer 3 > > > is never aware of it happening (although it may cause additional > delay). > > > Eventually the ethernet card will give up if too many collisions > occur > > > during retries, but this is very rare in practice unless the > network is > > > *VERY* loaded. > > > > > > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex > 10 meg > > > > ethernet would handle roughly 20-25 rtp sessions before bumping > into > > > > > > the > > > > > > > problem (your milage may vary). The majority of the folks on > this list > > > > seem to be running home/soho systems and would likely never run > into > > > > > > the > > > > > > > issue. But the heavier users will. > > > > > > For a duplex mismatch, my experience is that if one end on a 100 > Mb/sec > > > link is half and the other is full, bandwidth is limited to about > 8 > > > Mb/sec > > > max. This is based on some tests I've accidentally conducted. > If you > > > try > > > to send 9 Mb/sec over that link, yes, some packets will get > dropped as > > > they simply won't fit. (But I do agree that for a half-half > link, you > > > can > > > get about 20 Mb/sec) > > > > > > -- > > > Joel > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ---End of Original Message- > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/
Re: [Asterisk-Users] perl database get
On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote: > Does anyone have any example perl agi script that does a database > get. I am being thick and can't seem to get the return value: > > print "DATABASE PUT big bigger biggest \n"; This bit works > fine print "DATABASE GET big bigger \n"; > Now what do I do to get the my value from the database get?? $result = <>; or more explicitly: $result = ; The real answer, though, is to point you to the Perl module at http://asterisk.gnuinter.net and tell you that all of these issues have already been solved. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold in Conference room?
Hi! > Does anyone here know how I might provide music into a conference room > when there is only one participant. Dead silence tends to confuse > non-techies who think that they've done something wrong, even after the > entry announcement. MeetMe() now has an option M that does exactly that. Be sure to have configured music-on-hold (MOH) on your system. http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + CRM
Which events did you add ? Van: Jonathan Tew [mailto:[EMAIL PROTECTED] Verzonden: di 23/12/2003 16:25 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Asterisk + CRM We're starting to integrate * with our customer service software. Basically we're pulling off events from the management interface. We're also making some small patches to the code to deliver more events about the channel variables, etc. Anton Yurchenko wrote: > Hello, > > Anyone aware of any CRM products projects that intagrete with *? Or > that integrate with any telephony products? Is there some open API for > such integration, or are they all proprietory? > > Thanks > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users <>
Re: [Asterisk-Users] codes/grandstream/PRI.. few questions :)
Justin, Comments inline: At 01:06 24/12/03 +1100, you wrote: Hi Guys.. Just wondering if someone could help me with a few questions please. were currently using the ulaw codec with our grandstream/iconnect/asterisk setup and its working pretty good except for the fact it downloads heaps. Does anyone know a good site to get referances to how much each codec downloads/quality etc etc ? Ive tried using that g723 codec but i have have problems as soon as a i dial. my next question.. :) does anyone know howto fix the grandstream 484 errors you get sometimes when you dial ? i had a look at they rekon to put early dial on.. which just makes things worse heh. They'd be a cool little phone except for this problem. Lastly were looking at getting a PRI or something to handle 30 lines.. I know digium sells hardware to do this, has anyone in australia gotten good results from doing this kind of setup ?? also what are the restrictions in regards to caller id and that sort of stuff in aus? do is all work ? Haven't tested GS yet so I can't help you yet. There are a number using PRI here. We are awaiting final testing documents to be able to issue A Tick on TE410P-A. Have 'A tick' stickers ready to go to make official for Australia. You get caller id if the number isn't silent or hasn't requested a block on sending the caller id. thanks heaps everyone :) Merry Christmas Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Peter Brown CEO IP Telephonics ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] perl database get
I've used both the syntax you have given and the perl module. AGI->getvar() returns nothing for arguments that work from the CLI (Also when I run agi-test.agi, the only thing that works is the SAY NUMBER. SEND TEXT doesn't work and nothing at all is printed to teh console) I am using redhat 8. Could it be a redhat 8 problem do you think? - Original Message - From: "Tilghman Lesher" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, December 23, 2003 6:07 PM Subject: Re: [Asterisk-Users] perl database get > On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote: > > Does anyone have any example perl agi script that does a database > > get. I am being thick and can't seem to get the return value: > > > > print "DATABASE PUT big bigger biggest \n"; This bit works > > fine print "DATABASE GET big bigger \n"; > > Now what do I do to get the my value from the database get?? > > $result = <>; > or more explicitly: > $result = ; > > The real answer, though, is to point you to the Perl module at > http://asterisk.gnuinter.net and tell you that all of these issues > have already been solved. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Capi Dial & outgoing msn?
Hi!
> > try it without prefix (else dtag uses first msn) -
> > so if your city code is 07032 and phone no (msn) 41432
> > -> exten => _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
> >
> >
> > Thomas
>
> Thanks for the pointer Thomas. I removed the areacode from msn= in
> capi.conf and from the dial statement. Tried again and till no CLID.
> Stumped at this point. Perhaps my telco doesn't allow setting outgoing
> msn numbers.
If you read the CAPI documentation you'll find that @ will help you to
_hide_ your ID (this is called "CLIR") - however from your message I
understand that you want to do the opposite? So just drop the @.
If your problem persists: You might have told your telco to _always_ hide
your ID. Or maybe it's just that you need to remove the 0 before
703241432 as outgoing MSN.
Cheers, Philipp
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Fw: [Asterisk-Users] perl database get
i mean AGI->database_get() - Original Message - From: "Muhammad Nasim" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, December 23, 2003 6:41 PM Subject: Re: [Asterisk-Users] perl database get > I've used both the syntax you have given and the perl module. AGI->getvar() > returns nothing for arguments that work from the CLI > > (Also when I run agi-test.agi, the only thing that works is the SAY NUMBER. > SEND TEXT doesn't work and nothing at all is printed to teh console) > > I am using redhat 8. Could it be a redhat 8 problem do you think? > > - Original Message - > From: "Tilghman Lesher" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, December 23, 2003 6:07 PM > Subject: Re: [Asterisk-Users] perl database get > > > > On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote: > > > Does anyone have any example perl agi script that does a database > > > get. I am being thick and can't seem to get the return value: > > > > > > print "DATABASE PUT big bigger biggest \n"; This bit works > > > fine print "DATABASE GET big bigger \n"; > > > Now what do I do to get the my value from the database get?? > > > > $result = <>; > > or more explicitly: > > $result = ; > > > > The real answer, though, is to point you to the Perl module at > > http://asterisk.gnuinter.net and tell you that all of these issues > > have already been solved. > > > > -Tilghman > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] turning off IAX registration attempts
On Tuesday 23 December 2003 11:13, Robert Hajime Lanning wrote:
> I have, in iax.conf the register statement:
> register => username:[EMAIL PROTECTED]
>
> This causes registration attempts to iaxtel.com for both IAX and
> IAX2.
>
> Every once in a while there is a packet for port 4569 keeping the
> IAX2 registration alive. This is fine.
>
> But, I have a barrage of registration attempts to iaxtel on port
> 5036 for IAX. Every UDP packet is answered with an ICMP packet
> claiming "port unreachable".
>
> I know that iaxtel has turned off IAX, So, how do I turn off the
> registration attempts for IAX, for that particular connection?
> (and keep IAX2)
How's this for a solution (attached)?
-Tilghman
Index: channels/chan_iax.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_iax.c,v
retrieving revision 1.43
diff -u -r1.43 chan_iax.c
--- channels/chan_iax.c 9 Dec 2003 23:55:17 - 1.43
+++ channels/chan_iax.c 23 Dec 2003 18:43:41 -
@@ -4661,7 +4661,7 @@
} else if (!strcasecmp(v->value, "yes")) {
peer->maxms = DEFAULT_MAXMS;
} else if (sscanf(v->value, "%d", &peer->maxms) != 1) {
- ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of iax.conf\n", peer->name, v->lineno);
+ ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of iax1.conf\n", peer->name, v->lineno);
peer->maxms = 0;
}
} //else if (strcasecmp(v->name,"type"))
@@ -4962,7 +4962,7 @@
static int reload_config(void)
{
- char *config = "iax.conf";
+ char *config = "iax1.conf";
struct iax_registry *reg;
struct sockaddr_in dead_sin;
strncpy(accountcode, "", sizeof(accountcode)-1);
@@ -5359,7 +5359,7 @@
int load_module(void)
{
- char *config = "iax.conf";
+ char *config = "iax1.conf";
int res = 0;
int x;
struct iax_registry *reg;
[Asterisk-Users] Cisco 7960 phones.
I have found a phone that I wish I had not! This is by far the worst phone to setup. I have finally upgraded it to Sip but once this got done it I am not able to get it unlocked so I can enter the rest of the settings. So if anyone out there can tell me how to setup my DNS server to tell it where the tftp is located (Windows 2000 DNS server) or please let me have the default password. I called Cisco and they said I can change it with the tftp setup and that is all they said. (They also said to use Skinny and not SIP) Not much help from there support contract we have on with them! Why have they made a phone that is so hard to get working! I feel that Cisco does not understand the KISS format! I have read the Wiki and the setup is listed there incorrectly as 9740/9760 instead of 7940/7960 (Can someone rename that please). Thank you all for allowing to do a little venting as well. Sorry did not mean to do this! But this has gotten me to loose hair and I don't have much left! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] please help - ztdummy problems
I have read a lot about ztdummy, but I miss something. I don't have any digium hardware, but want to do Meetme. I read that I need ztdummy installed in order to do a conference room. I followed all steps to get ztdummy compiled and installed (including uncoment on makefile) When I install the module, my * sound becomes unrecognizably choppy on every channel type, not only meetme! So I have been forced to uninstall the module to return my * to an usable state. Anybody can help me to solve this? Thanks a lot. Hector. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Capi Dial & outgoing msn?
On Tue, 2003-12-23 at 19:39, Philipp von Klitzing wrote: [snip] > If you read the CAPI documentation you'll find that @ will help you to > _hide_ your ID (this is called "CLIR") - however from your message I > understand that you want to do the opposite? So just drop the @. > > If your problem persists: You might have told your telco to _always_ hide > your ID. Or maybe it's just that you need to remove the 0 before > 703241432 as outgoing MSN. > > Cheers, Philipp > Hi Philipp, The text in the chan_capi README sort of tells you to use it: "Using CLIR == in the Dial command put a '@' infront of the msn you want to use for dialing out, e.g.: s,1,Dial,CAPI/@12345678:BYEXTENSION|30|r" My interpretation of that text is that I need to use the "@" so I can set the outgoing MSN. If I had known that the R in CLIR means something like Restrict it would have been obvious :) Luckily your pointer solved my problem so thanks for that. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 phones.
What firmware did you upgrade to? If its version 5.0 and above the default password is cisco and to unlock it you press settings then 9. NO cisco's docs are simple.. You are just trying too hard. bkw On Tue, 23 Dec 2003, Ariel Batista wrote: > I have found a phone that I wish I had not! This is by far the worst > phone to setup. I have finally upgraded it to Sip but once this got > done it I am not able to get it unlocked so I can enter the rest of the > settings. So if anyone out there can tell me how to setup my DNS server > to tell it where the tftp is located (Windows 2000 DNS server) or please > let me have the default password. I called Cisco and they said I can > change it with the tftp setup and that is all they said. (They also said > to use Skinny and not SIP) Not much help from there support contract we > have on with them! Why have they made a phone that is so hard to get > working! I feel that Cisco does not understand the KISS format! I have > read the Wiki and the setup is listed there incorrectly as 9740/9760 > instead of 7940/7960 (Can someone rename that please). > > Thank you all for allowing to do a little venting as well. Sorry did > not mean to do this! But this has gotten me to loose hair and I don't > have much left! > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voiceglo setup for home
I am looking to speak to anyone else that has connected to Voiceglo using Asterisk. I'm using SIP and have most of the issues worked out. But remote outbound ringing doesn't work. So it would be nice to discuss configs. Maybe someone out there is using IAX instead. cameron. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 phones.
Brian West wrote: > What firmware did you upgrade to? > > If its version 5.0 and above the default password is cisco and to > unlock it you press settings then 9. > > NO cisco's docs are simple.. You are just trying too hard. I want to thank you for the password of cisco. It worked. I have finally gotten the phone to work. Now to start setting all the new bells and tones it has! I still think that they have over done the settings on the phone! > bkw > > On Tue, 23 Dec 2003, Ariel Batista wrote: > >> I have found a phone that I wish I had not! This is by far the worst >> phone to setup. I have finally upgraded it to Sip but once this got >> done it I am not able to get it unlocked so I can enter the rest of >> the settings. So if anyone out there can tell me how to setup my >> DNS server to tell it where the tftp is located (Windows 2000 DNS >> server) or please let me have the default password. I called Cisco >> and they said I can change it with the tftp setup and that is all >> they said. (They also said to use Skinny and not SIP) Not much help >> from there support contract we have on with them! Why have they >> made a phone that is so hard to get working! I feel that Cisco does >> not understand the KISS format! I have read the Wiki and the setup >> is listed there incorrectly as 9740/9760 instead of 7940/7960 (Can >> someone rename that please). >> >> Thank you all for allowing to do a little venting as well. Sorry did >> not mean to do this! But this has gotten me to loose hair and I >> don't have much left! >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 phones.
I've got to agree. Once you figure out the first phone, all the others take about 30 seconds to configure. The Cisco SIP documentation is located at: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/index.htm cameron. On Tue, 23 Dec 2003, Brian West wrote: > What firmware did you upgrade to? > > If its version 5.0 and above the default password is cisco and to unlock > it you press settings then 9. > > NO cisco's docs are simple.. You are just trying too hard. > > bkw > > On Tue, 23 Dec 2003, Ariel Batista wrote: > > > I have found a phone that I wish I had not! This is by far the worst > > phone to setup. I have finally upgraded it to Sip but once this got > > done it I am not able to get it unlocked so I can enter the rest of the > > settings. So if anyone out there can tell me how to setup my DNS server > > to tell it where the tftp is located (Windows 2000 DNS server) or please > > let me have the default password. I called Cisco and they said I can > > change it with the tftp setup and that is all they said. (They also said > > to use Skinny and not SIP) Not much help from there support contract we > > have on with them! Why have they made a phone that is so hard to get > > working! I feel that Cisco does not understand the KISS format! I have > > read the Wiki and the setup is listed there incorrectly as 9740/9760 > > instead of 7940/7960 (Can someone rename that please). > > > > Thank you all for allowing to do a little venting as well. Sorry did > > not mean to do this! But this has gotten me to loose hair and I don't > > have much left! > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] perl database get
On Tuesday 23 December 2003 12:41, Muhammad Nasim wrote: > I've used both the syntax you have given and the perl module. > AGI->getvar() returns nothing for arguments that work from the CLI Try AGI->get_variable() > (Also when I run agi-test.agi, the only thing that works is the SAY > NUMBER. SEND TEXT doesn't work and nothing at all is printed to teh > console) SEND TEXT does not print anything on the console. SEND TEXT sends text on the channel, if the channel supports it. Audio-only channels do not support the SEND TEXT command. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] perl database get
On Tuesday 23 December 2003 12:44, Muhammad Nasim wrote: > i mean AGI->database_get() Then that probably means that the database key does not exist. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] Fw: Questions and finding
> Thanks for the reply. > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. I > am not sure if OutboundProxy has to be configured to have it working fine. > Or this just happened to me? What is your ATA's software? > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > As per ATA, it is by default using rfc2833. I tried setting it up as inband > by setting Audiomode, but nothing helped. I was thinking the * is ONLY > recognizing the DTMF if there is telco board installed. Is it? > > > - Original Message - > From: "Philipp von Klitzing" <[EMAIL PROTECTED]> > To: "Jess Magnaye" <[EMAIL PROTECTED]> > Sent: Tuesday, December 23, 2003 12:36 PM > Subject: Re: [Asterisk-Users] Fw: Questions and finding > > > > Hi! > > > > > 1.) First test > > > - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off > > > after 5-10seconds (consistently). > > > - Solution: I have to reconfigure ATA to use OutboundProxy to be > Asterisk > > > IP. > > > - Am I doing the right thing? > > > > Turn of silence detection / VAD. > > > > > Any solution to this one? > > > My thinking was that DTMF can only be detected by * > > > > Take a look at your SIP configuration and make sure you have the correct > > dtmfmode= set. Try different values if you continue to have trouble and > > configure your ATA accordingly. > > > > Cheers, Philipp > > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
How do you reset the unit without pulling out the plug. The easiest way to get the info you are looking for, is to get an 8 buck CCO account. On Tue, 23 Dec 2003, Adthrawn wrote: > Hi, > > Has anybody been successful in running the 7914 expansion unit for the > Cisco 7960G IP phone? For anybody unaware of what the expansion unit > does, it provides 14 additional buttons, with an LCD display. The idea, > is that with an expansion unit (a 7960 can take upto 2 of these units), > a user can either assign more speed-dial's, or can monitor line > status/account status. So, you can either register a speed-dial or > register another account. > > The problem I've found so far, is that speed-dials are not programmed > on the phone, but are instead handled by the Call Manager software (not > on a user basis, but on a phone, MAC address basis). Likewise, plugging > the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red > lights (the buttons also light-up red, blue or green), which according > to the hidden technical documentation, indicates that the Call Manager > is not registering the unit. I can't work out if it's short of firmware > embedded in the Call Manager, whether it's searching for a > configuration file on the TFTP (Cisco phones need a TFTP to get their > settings and SIP firmware), whether it's not happy with the phone being > a SIP version, or whether I'm doing something wrong. > > I've had to learn about the 7960's configuration the hard way, and > despite their useless technical documents, have managed to configure > most settings. > > There's quite a bit of extra configuration for the 7960 I'd love to get > to, and would like help or advice on. Things like directory services, > screen logo, the 7914 and more! > > If anybody is interested, I have resources and files to; convert from > Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail > indicator lamp, special key combinations to reset the unit (without > pulling the plug out) and locking/unlocking the preferences, > configuring the voicemail speed-dial > > Any help or advice, please let me know! > > Regards, > Ad. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Questions and finding
> Hi! > > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. > > Then check if you have a firewall in between * and your ATA that closes > the port due to inactivity of your ATA. Also use "SIP DEBUG" in the CLI > to try to see a bit more of what is going on. You could also use Ethereal > to monitor the SIP traffic (or the rtp/UDP traffic). > > > am not sure if OutboundProxy has to be configured to have it working fine. > > Or this just happened to me? What is your ATA's software? > > I don't have such a device, in fact never had. :-) MY ATA and * are sitting on the same LAN. So FW or NAT problem is not possible. This is also the reason why I commented out "nat=1" in the sip.conf. > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > > Note: inband only works with g.711 codec. Doesn't the ATA also offer > "info" as third dtmfmode option? Anyway, you might want to search the > mailing list for setup info, there are a lot of people around that use > it. > > > by setting Audiomode, but nothing helped. I was thinking the * is > > ONLY recognizing the DTMF if there is telco board installed. Is it? > > No no, * doesn't require any hardware to be installed. > LET ME TRY dtmfmode=info AND SEE WHAT HAPPENS NEXT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Starting Asterisk
Hi Chris, In this situation, how do I modprobe ztdumy before * get started ? SW Message: 6 Date: Tue, 23 Dec 2003 09:33:07 -0800 (PST) From: Chris Albertson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Auto Starting Asterisk To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Look in the directory /etc/init.d (/etc/rc.d/init.d on some systems) You put a script in there called "asterisk". There is a sample called "asterisk.init" in the source. copy it to /etc/init.d/asterisk You may want to study the other files in /etc/init.d to see how they work. Next read the "chkconfig" man page and see way you'd want to type "chkconfig --add asterisk; chkconfig asterisl on" Finally to start asterisk you can type "./asterisk start" You may also want to re-boot the computer to verify that asterisk does start automatically ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
7914's don't work with SIP. SCCP only. And why do people keep talking about this 8 dollar CCO account ... Its a service contract on the Cisco ATA-186. The one for the 79XX's are over 80.00/yr bkw On Tue, 23 Dec 2003, Lists wrote: > How do you reset the unit without pulling out the plug. The easiest way > to get the info you are looking for, is to get an 8 buck CCO account. > > > On > Tue, 23 Dec 2003, Adthrawn wrote: > > > Hi, > > > > Has anybody been successful in running the 7914 expansion unit for the > > Cisco 7960G IP phone? For anybody unaware of what the expansion unit > > does, it provides 14 additional buttons, with an LCD display. The idea, > > is that with an expansion unit (a 7960 can take upto 2 of these units), > > a user can either assign more speed-dial's, or can monitor line > > status/account status. So, you can either register a speed-dial or > > register another account. > > > > The problem I've found so far, is that speed-dials are not programmed > > on the phone, but are instead handled by the Call Manager software (not > > on a user basis, but on a phone, MAC address basis). Likewise, plugging > > the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red > > lights (the buttons also light-up red, blue or green), which according > > to the hidden technical documentation, indicates that the Call Manager > > is not registering the unit. I can't work out if it's short of firmware > > embedded in the Call Manager, whether it's searching for a > > configuration file on the TFTP (Cisco phones need a TFTP to get their > > settings and SIP firmware), whether it's not happy with the phone being > > a SIP version, or whether I'm doing something wrong. > > > > I've had to learn about the 7960's configuration the hard way, and > > despite their useless technical documents, have managed to configure > > most settings. > > > > There's quite a bit of extra configuration for the 7960 I'd love to get > > to, and would like help or advice on. Things like directory services, > > screen logo, the 7914 and more! > > > > If anybody is interested, I have resources and files to; convert from > > Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail > > indicator lamp, special key combinations to reset the unit (without > > pulling the plug out) and locking/unlocking the preferences, > > configuring the voicemail speed-dial > > > > Any help or advice, please let me know! > > > > Regards, > > Ad. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conf file system generation in * for User/Admin update
Is there anyone who could show me code (or point me in the right direction) to allow users or PABX Admin to generate their own * conf files. If there isn't anything I will just have to start it myself. Any suggestions for basics to start with. I believe the issues are going to be about dependencies between the various conf. files. Any help would be gratefully received. A GUI for this would be great or just via a web browser. __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Questions and finding
Hi Philip, I found the problem. My sip.conf config was changed by somebody else. :( The external IP was uncommented and that's what is causing my problem. - Original Message - From: "Jess Magnaye" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Tuesday, December 23, 2003 4:33 PM Subject: Re: [Asterisk-Users] Fw: Questions and finding > > > > Hi! > > > > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. > > > > Then check if you have a firewall in between * and your ATA that closes > > the port due to inactivity of your ATA. Also use "SIP DEBUG" in the CLI > > to try to see a bit more of what is going on. You could also use Ethereal > > to monitor the SIP traffic (or the rtp/UDP traffic). > > > > > am not sure if OutboundProxy has to be configured to have it working > fine. > > > Or this just happened to me? What is your ATA's software? > > > > I don't have such a device, in fact never had. :-) > > MY ATA and * are sitting on the same LAN. So FW or NAT problem is not > possible. This is also the reason why I commented out "nat=1" in the > sip.conf. > > > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None > worked. > > > > Note: inband only works with g.711 codec. Doesn't the ATA also offer > > "info" as third dtmfmode option? Anyway, you might want to search the > > mailing list for setup info, there are a lot of people around that use > > it. > > > > > by setting Audiomode, but nothing helped. I was thinking the * is > > > ONLY recognizing the DTMF if there is telco board installed. Is it? > > > > No no, * doesn't require any hardware to be installed. > > > LET ME TRY dtmfmode=info AND SEE WHAT HAPPENS NEXT. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Sounds patchy.
I have gotten the Cisco 7960 working with my Asterisk system under SIP. The version is 5.03 that I am using. Cisco Support said I should not upgrade to version 6 yet. My next question is the sound is patchy when people here me. But I can hear them just fine not patchy. I have the 188 page Admin manual and it seem not to say anything about improving the sound. All other phones like IPDialog work fine without the patchy sound. I have tried ulaw and alaw as the codex. Both sound the same! Is there any other settings that can be done. I remember that the X-lite has a transmit silence but I could not find this setting in there documentation. P.S. the Contract for the 7960 cost us $ 83.40 for each phone. This I feel is high. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8 Minus the DTA
I know someone mentioned doing this once before however I can’t find it. Anyone remember if or how it was successful? Thanks!
Re: [Asterisk-Users] Cisco 7960 Sounds patchy.
Ariel Batista wrote: P.S. the Contract for the 7960 cost us $ 83.40 for each phone. This I feel is high. This smells like a Cisco re-certification fee to me. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs
resending. Can anyone help me in trying to understand what would be the problem. appreciate ur time. i need to get this working. thanks a lot, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 8:15 PM Subject: [Asterisk-Users] MSN to GS - Call drops in 10 secs Hi All, i dont know what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general]port = 5060bindaddr = 0.0.0.0context = bogon-calls;context = defaultdisallow=allallow=ulawallow=alawallow=ilbcallow=gsm ;My SIP phone - GS[2000]type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband ;MSN Msgr[2002]type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji Do you Yahoo!?Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
[Asterisk-Users] Merry Christmas, all Asterisk users!
It's the day before Christmas here in Sweden, actually the night before at this time... We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into "merry-christmas-mode" with the yet undocumented CLI command "frosty-mode on", a mode where the PBX will connect all incoming extensions to the "ho-ho-ho" sound file and then randomly pick a number in the +1234 country code (for the North Pole), dial out and bridge. And these magical SIP connections will work over ANY type of NAT. (Due to the SIP header "Santa-magic-cookie: on") And yes, the frosty mode is even un-documented on the http://www.voip-info.org wiki. :-) It's been fun spending the fall with the Asterisk project. I look forward to next year, with the new handbook coming in place, with many new applications and features and - hopefully - many new Asterisk installs at customer sites. It's snowing outside, the trees are already covered with snow and the stars are glittering on a dark sky. My kids are sleeping, dreaming of their christmas gifts tomorrow. It's going to be a traditional Swedish christmas... Have a wonderful Christmas, all of you! Warm regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP / FXS - MOH
Is there anway to do MOH on a FXS extension like what is done using SIP. There has to be a way within manager or something, to send this call to MOH and then retreive the call. I need to set this up, so users are just hitting one button to put callers on hold and one or another button to retrieve the users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] configuration files for cisco 7960
Is there any place where I can download sample files for the cisco 7960 (SIP) ?
[Asterisk-Users] Voiceglo SIP configuration
The call quality is really pretty good. I think better than Vonage over
an FXO bridge. If you are looking for a home provider with direct SIP
support and local phone numbers this is a good choice. If anyone has
questions or comments about my configuration please pass them along. I
have noticed that if you don't put fromuser=phone# then the extension
caller id passes through. Also the major annoyance is why outbound
calling gives no ring indication. I'm still looking into whether there is
no ring indicator being sent back, or how to create one. Using the little
'r' at the end of the dial string just seems to prevent the call from
going through.
Username is your 10-digit phone number.
Password is in the .reg file they sent you via email. I signed up for the
USB phone, so I don't if they send a .reg file if you went for the MTA.
sip.conf
register => 1234567890:[EMAIL PROTECTED]
[myphone.voiceglo.com]
type=peer
username=1234567890
secret=password
nat=no
host=myphone.voiceglo.com
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=200
restrictid=no
fromuser=1234567890
fromdomain=1234567890.voiceglo.com
extensions.conf
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
cameron.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merry Christmas, all Asterisk users!
Merry Christmas Ollie from all of us Asterisk people in the US/East Coast region. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
If you purchase a new telephone, the warranty is more like $15. It's more for used phones. Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, December 23, 2003 2:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials 7914's don't work with SIP. SCCP only. And why do people keep talking about this 8 dollar CCO account ... Its a service contract on the Cisco ATA-186. The one for the 79XX's are over 80.00/yr bkw On Tue, 23 Dec 2003, Lists wrote: > How do you reset the unit without pulling out the plug. The easiest way > to get the info you are looking for, is to get an 8 buck CCO account. > > > On > Tue, 23 Dec 2003, Adthrawn wrote: > > > Hi, > > > > Has anybody been successful in running the 7914 expansion unit for the > > Cisco 7960G IP phone? For anybody unaware of what the expansion unit > > does, it provides 14 additional buttons, with an LCD display. The idea, > > is that with an expansion unit (a 7960 can take upto 2 of these units), > > a user can either assign more speed-dial's, or can monitor line > > status/account status. So, you can either register a speed-dial or > > register another account. > > > > The problem I've found so far, is that speed-dials are not programmed > > on the phone, but are instead handled by the Call Manager software (not > > on a user basis, but on a phone, MAC address basis). Likewise, plugging > > the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red > > lights (the buttons also light-up red, blue or green), which according > > to the hidden technical documentation, indicates that the Call Manager > > is not registering the unit. I can't work out if it's short of firmware > > embedded in the Call Manager, whether it's searching for a > > configuration file on the TFTP (Cisco phones need a TFTP to get their > > settings and SIP firmware), whether it's not happy with the phone being > > a SIP version, or whether I'm doing something wrong. > > > > I've had to learn about the 7960's configuration the hard way, and > > despite their useless technical documents, have managed to configure > > most settings. > > > > There's quite a bit of extra configuration for the 7960 I'd love to get > > to, and would like help or advice on. Things like directory services, > > screen logo, the 7914 and more! > > > > If anybody is interested, I have resources and files to; convert from > > Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail > > indicator lamp, special key combinations to reset the unit (without > > pulling the plug out) and locking/unlocking the preferences, > > configuring the voicemail speed-dial > > > > Any help or advice, please let me know! > > > > Regards, > > Ad. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN messenger and *
Speaking of MSN/Windows Messenger, how does one call someone? Using the configuration specified, I've registered it with Asterisk, but it requires that I add a Passport contact. Does anyone have experience calling a sip endpoint without it being a Passport account? -g On Mon, 2003-12-22 at 20:42, Balaji NJL wrote: > use this > > > [3001] > > type=friend > > ;username=3001 > > ;fromuser=Craig1 > > ;secret=secret > > host=dynamic > > mailbox=3001 > > context=sip > > dtmfmode=info > > auth=plaintext > > > > make sure ur MSN version is 4.7.0105. > > > > -B > > > - Original Message - > From: Craig Waddington > To: [EMAIL PROTECTED] > Sent: Monday, December 22, 2003 10:10 AM > Subject: [Asterisk-Users] MSN messenger and * > > > Sorry for the late reply. > > > > I try port 5060 and it just knocks me back straight away, I > cant see it even try to authenticate in the CLI. > > > > X-lite works both inside the LAN and outside using SIP. > > > > Messenger version = 4.7 > > > > John I will try your suggestion with sip.conf thanks for the > help. I notice a few differences, I seem to be missing some > bits.. > > > > Its like it is trying to authenticate with the Linux box and > not asterisk. > > > > Sip.conf > > > > [general] > > port=5060 ; Port to bind to > > bindaddr=0.0.0.0; Address to bind to > > context=sip ; Default for incoming calls > > allow=ulaw > > allow=alaw > > allow=gsm > > allow=ilbc > > > > > > [3001] > > type=friend > > username=3001 > > fromuser=Craig1 > > secret=secret > > host=dynamic > > mailbox=3001 > > context=sip > > dtmfmode=info > > > > I found 3 guides and each one seems to be a bit different and > use different ports. > > > > I am using the X100P, it is a home system, to reduce call > charges for my family overseas. > > > > If I can get Messengger working it will be easier to talk > them through the setup. > > > > > > > > > > __ > Do you Yahoo!? > Yahoo! Photos - Get your photo on the big screen in Times Square ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MSN to GS - Call drops in 10 secs
Balaji, I also have the same issue. Works fine on any phone except GS for me. After a bit of research I found a post saying set the phone to “offer only one codec set”. It looks like we have to set the phone to use one codec – GSM I am concerned that you cant use passwords when logging in to * using Messenger. Craig. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 23 December 2003 23:04 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs resending. Can anyone help me in trying to understand what would be the problem. appreciate ur time. i need to get this working. thanks a lot, -B - Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 8:15 PM Subject: [Asterisk-Users] MSN to GS - Call drops in 10 secs Hi All, i dont know what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS [2000] type=friend username=2000 secret=qweqwe host=dynamic context=from-sip mailbox=2000 dtmfmode=inband ;MSN Msgr [2002] type=friend host=dynamic insecure=yes dtmfmode=inband ;dtmfmode=rfc2833 context=from-sip mailbox=2002 ;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
100% agree. I think this thread is getting strung out much further then Olle's original question relative to commenting on half vs full duplex. Lots of great discussion though thanks to all that participated! Rich > The reason you use UDP over TCP for realtime meadia is that > TCP's ability to reliably deliver every packet in order actually > sounds worse. Reason being is that with a UDP system a dropped > packet sounds like just a dropout but if you used TCP the audio > stream would be held up and delayed in a queue while that lost > packet was being retransmitted. In stead of a dropout the audio > would sound as if someone kept hitts a "pause" button on a tape > recorder. A dropout sounds better then a delay of potentialy > several seconds > > Almost all realtime meadia systems (telephony, video, possition > reporting and so on) maintain some kind of a buffer on the recieving > end. But you trad the buffer lenght for delay. Using UDP allows > the application to do the buffering where as TCP putting this buffing > functin in the operaing systems network code. > > > > --- Andres <[EMAIL PROTECTED]> wrote: > > On Tuesday 23 December 2003 11:40, Rich Adamson wrote: > > > There's no reassembly with udp, and there is no sense of packets > > arriving > > > in the same order as what was sent. Udp is a best-effort > > low-overhead way > > Right, UDP itself does not care about order, but at the application > > layer you > > can keep track of it. You can design your application to buffer X > > packets > > and then reorder them according to sequence numbers. > > > > > of transmitting data (with UDP often times referred to as the > > Unreliable > > > Data Protocol). Changing to TCP would allow reassembly, however the > > > overhead would be substantial. > > > > > > > > > > > > > The problem occurs when the software is expecting the packet in a > > certain > > > > timeframe so that it can reassemble it in a timely manner. It's > > not a > > > > big deal with a web page or something along that lines. But when > > a voice > > > > application cannot get reassembled in a timely manner, you'll > > surely > > > > notice it! > > > > > > > > -Original Message- > > > > From: Joel Maslak > > > > To: [EMAIL PROTECTED] > > > > Sent: 12/23/2003 10:41 AM > > > > Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) > > > > > > > > On Tue, 23 Dec 2003, Rich Adamson wrote: > > > > > If a collision or dropped packet occurs (in a voip udp > > environment) > > > > > > > > there > > > > > > > > > is no way to retransmit the missing/damaged packet. Missing one > > packet > > > > > > > > isn't > > > > > > > > > a big deal, but if you have collisions and/or dropped packets, > > there > > > > > > > > is a > > > > > > > > > very high probability that lots of packets will be dropped. If > > too > > > > > > > > many > > > > > > > > > are dropped, you'll hear the result in the undecoded voice as > > choppy > > > > > voice. > > > > > > > > Actually, collisions occur at Layer 2, not Layer 3, and the layer > > 2 > > > > hardware automatically resends packets involved in a collision - > > layer 3 > > > > is never aware of it happening (although it may cause additional > > delay). > > > > Eventually the ethernet card will give up if too many collisions > > occur > > > > during retries, but this is very rare in practice unless the > > network is > > > > *VERY* loaded. > > > > > > > > > Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex > > 10 meg > > > > > ethernet would handle roughly 20-25 rtp sessions before bumping > > into > > > > > > > > the > > > > > > > > > problem (your milage may vary). The majority of the folks on > > this list > > > > > seem to be running home/soho systems and would likely never run > > into > > > > > > > > the > > > > > > > > > issue. But the heavier users will. > > > > > > > > For a duplex mismatch, my experience is that if one end on a 100 > > Mb/sec > > > > link is half and the other is full, bandwidth is limited to about > > 8 > > > > Mb/sec > > > > max. This is based on some tests I've accidentally conducted. > > If you > > > > try > > > > to send 9 Mb/sec over that link, yes, some packets will get > > dropped as > > > > they simply won't fit. (But I do agree that for a half-half > > link, you > > > > can > > > > get about 20 Mb/sec) > > > > > > > > -- > > > > Joel > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ---End of Original Message- > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] >
Re: [Asterisk-Users] Cisco 7960 phones.
> > What firmware did you upgrade to? > > > > If its version 5.0 and above the default password is cisco and to > > unlock it you press settings then 9. > > > > NO cisco's docs are simple.. You are just trying too hard. > > I want to thank you for the password of cisco. It worked. I have > finally gotten the phone to work. Now to start setting all the new > bells and tones it has! I still think that they have over done the > settings on the phone! For those of us that have been around the block (at least a couple of times), the quality of the Cisco product (produced by some other company that I can't seem to remember ;) ) is significantly greater then a number of other devices, once you understand some of the values that aren't noticed in 10 seconds or less. Enjoy (... I don't sell/work for/sponsor Cisco products) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CT1 and callerid
I'm just double checking.. I was told it wasn't possible but i'm going to ask just in case. Can you set outbound callerid on a channelized T1? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merry Christmas, all Asterisk users!
> It's the day before Christmas here in Sweden, actually the night before > at this time... > It's been fun spending the fall with the Asterisk project. I look forward > to next year, with the new handbook coming in place, with many new > applications and features and - hopefully - many new Asterisk installs at > customer sites. > Have a wonderful Christmas, all of you! And from all of us that have been around this list for a while, we Thank YOU for taking the time and effort placed towards advancing the documentation and participation. You are "the man"! May the almighty one grant speed to the wiki! :) (Absolutely no offence intended; from one US swed to another swed!) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CT1 and callerid
On Tue, 2003-12-23 at 19:22, Brian West wrote: > I'm just double checking.. I was told it wasn't possible but i'm going to > ask just in case. > > Can you set outbound callerid on a channelized T1? I think there is a way to do something like DID with the 4 digits of DTMF passed before the call. It is unlikely though that you will find someone interested in doing that though. It is easier/cheaper to drop a PRI into somewhere and then outbound caller ID isn't kludgey with DTMF. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
> 7914's don't work with SIP. SCCP only. And why do people keep talking > about this 8 dollar CCO account ... Its a service contract on the Cisco > ATA-186. The one for the 79XX's are over 80.00/yr Careful Brian... things aren't always what they seem. There is some flexibility built into their P/L plan! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CT1 and callerid
HAHA you apparenlty aren't where we are... PRI is over priced... 3600/mth SBC Victim... they have to backhaul it 110 miles.. where CT1's can be served by the local CO. bkw On Tue, 23 Dec 2003, Steven Critchfield wrote: > On Tue, 2003-12-23 at 19:22, Brian West wrote: > > I'm just double checking.. I was told it wasn't possible but i'm going to > > ask just in case. > > > > Can you set outbound callerid on a channelized T1? > > I think there is a way to do something like DID with the 4 digits of > DTMF passed before the call. It is unlikely though that you will find > someone interested in doing that though. It is easier/cheaper to drop a > PRI into somewhere and then outbound caller ID isn't kludgey with DTMF. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Sounds patchy.
> I have gotten the Cisco 7960 working with my Asterisk system under SIP. > The version is 5.03 that I am using. Cisco Support said I should not > upgrade to version 6 yet. My next question is the sound is patchy when > people here me. But I can hear them just fine not patchy. I have the > 188 page Admin manual and it seem not to say anything about improving > the sound. All other phones like IPDialog work fine without the patchy > sound. I have tried ulaw and alaw as the codex. Both sound the same! > Is there any other settings that can be done. I remember that the > X-lite has a transmit silence but I could not find this setting in there > documentation. Patchy sound has absolutely nothing to do with the 7960 software. All versions from 2.1 through 6.0 have been solid as a rock from a usability standpoint. If you have patchy sound, look towards improper configuration of the phone vs asterisk definitions. In my (somewhat biased) opinion, the 7960 is "the" top of the line once you understand sip/rtp basics and provide the network infrastructure to support the basics. (It's also the most expensive even for refurb 7960's. Your milage may vary.) I avoided the Cisco v5 code for some time due to the back-level warnings published by Cisco. However, I upgraded to v6.0 code and would recommend it at a heart beat. It's been absolutely solid, stable, and they finally added some rather useful "user" features. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) & Help With 7960's Speed-dials
The fun part is getting a clueful reseller on the phone to sell you the correct thing. bkw On Tue, 23 Dec 2003, Rich Adamson wrote: > > 7914's don't work with SIP. SCCP only. And why do people keep talking > > about this 8 dollar CCO account ... Its a service contract on the Cisco > > ATA-186. The one for the 79XX's are over 80.00/yr > > Careful Brian... things aren't always what they seem. There is some > flexibility built into their P/L plan! > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outdialing with Voicetronix
Hi all,
Just thought I'd pass along some pointers when outdialing with Voicetronix's
OpenLine4 card.
I was having a tough time dialing out from *, it probably has something to
do with chan_vpb.c not waiting to hear the dialtone before telling the card
to dial. A quick fix was to insert a "," in the dialstring telling the card
to pause before dialing.
However when the "," was used in the dialstring, the Dial application
interpreted this as a command separator and was screwing things up. What I
did was to define
OUTDIAL=vpb/1-1/,,55
and used
exten => 555,s,1,Dial(${OUTDIAL})
to place the call. Works like a charm. Hopefully this bit of info may help
other VPB users out there.
Cheers,
Faiz
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[Asterisk-Users] Merry Christmas!
Merry Christmas from the Colorado Organization for Victims' Assistance. Our (Comdial) PBX fried after a power failure. Thanks to Mark Spencer, Digium, VCCH, and the friends who support this group, we are now back "on the air". We wish everyone good health for the coming year. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
