[Asterisk-Users] Remotely rebooting Sipura SPA-3000 from command line
Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:[EMAIL PROTECTED]/admin/reboot The strange thing is it works fine when I go to http://admin:[EMAIL PROTECTED]/admin/reboot with my web browser... Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remotely rebooting Sipura SPA-3000 from command line
On Sat, Aug 13, 2005 at 04:29:07PM +1000, Eric Bishop wrote: Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:[EMAIL PROTECTED]/admin/reboot The strange thing is it works fine when I go to http://admin:[EMAIL PROTECTED]/admin/reboot with my web browser... The sipura web interface seems to involve some very unnecessary javascript which makes it unusable from a textual browser. First thing to try is to disable javascript. Second thing: go there directly (not as part of a session). Maybe also from a freshly-opened browser. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: call load balancing
Hi Kind of late reeply. Still it mat be useful for others. On Wed, Aug 10, 2005 at 05:07:42PM +0100, Kevin Walsh wrote: Joseph [EMAIL PROTECTED] wrote: Don't forget to experiment with nice to increase priority of for Asterisk. By default asterisk run with priority 0 same as apache and any other applications. Actually, nice level is not exactly a priority. It is mearly a priority bonus. More on that below. There are two ways I can think of: 1. use nice in the command that runs asterisk. If using start-stop-daemon, Be sure to leave the --exec parapeter in tact and add --startas. Another option is to use renice: renice -10 -u asterisk or: renice -15 -p `cat /var/run/asterisk/asterisk.pid` We run a web-server on the same machine as asterisk and increasing nice for Asterisk to -15 helped a lot. You don't need to mess about with nice. Just run Asterisk with realtime priority; Use the -p switch when you start the Asterisk daemon. -p puts Asterisk in real-time priority. This is much more aggressive than the nice level. When a number of processes are ready to run, the system's scheduler needs to choose which of them will run. A common technique on most OSes is to keep a number that indicates how much is the process entitled to use the CPU. Whenever the process uses the CPU the number is increased (at least in unix) and only slowly reduced over time. A nice level changes that base priority. It puts the process still on the smae playing field with the other processes, but only gives it a small boost. Real-time priority[1] is something different: if the schduler sees proccesses with real-time priorities that have something to do, then they will do it. This is generally what we would want, because we would want Asterisk to handle sound fast. However: this means that a bug that makes Asterisk consume 100%CPU will practically stall the system. The system will still answer pings and open sockets, because that is done in the kernel. But nothing further will be done. /methinks are there threads of Asterisk that don't need to run in real-time priority? Are there parts of Asterisk that don't need to run in in real-time priority that could be moved to separate lower-priority threads? Call this priv-separation :-) . [OK, OK, don't ask -devel questions about post-1.2 on the -users list. I know :-) ] [1] Not to be confused with real-time configuration query from an external source, which is the normal meaning of real-time in Asterisk nowadays. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] real-time priority , -p switch
On Fri, Aug 12, 2005 at 09:47:21AM -0600, Joseph wrote: Though is the way to verify that asterisk is running with -p switch? If Asterisk has failed to get real-time priority it should print an appropriate error message and exit. I've modified the startup script to start asterisk with -p; however asterisk starts several sub-precess-ID's. Do the sup-process-ID's are effected by the -p switch? They are all threads and children of the same process. Unless there was an explicit proirity change somewhere along the way. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wildcard/FXO config
On Thu, Aug 11, 2005 at 11:20:41PM -0500, David Williams wrote: Trying to config the latest Asterisk/zaptel with an Digium Wildcard and a single X100m FXO interface connected to a POTS analog line. Build and install of both work ok - I'm using Suse 8 on a dual Pentium box. I load the driver with modprobe wctdm and the LED on the wildcard lights up. Then I start Asterisk with asterisk -vvvgc and asterisk fails to start. The messages look pretty good till the last few lines.. Shouldn't wcfxo be used for X100P (and clones)? On zaptel 1.0 wctdm is simply an alias to wcfxs, right? What is the output of: lsmod | grep zaptel cat /proc/zaptel/* -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fc3 build after kernel update?
On Fri, Aug 12, 2005 at 09:20:54PM -0600, Rich Adamson wrote: Updated the kernel on a fc3 box from 2.6.9-1.667 to 2.6.12-1.1372_FC3 today. Now the cvs-head for zaptel won't compile (libpri and asterisk does). The problem seems to be a symlink issue with the zaptel/Makefile looking for: KSRC:=/lib/modules/$(KVERS)/build So work around the problem: make KSRC=path/to/your/kernel/source/tree If there is any other place where the kernel source tree is referred to in the makefile it is a bug. Speaking of bugs: http://bugs.digium.com/view.php?id=4355 I'm still very unhappy with the install target there. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incompatible destination (88) Error Message
Hi, What do you mean by saying Bearer Capability. Either Speech, or 3.1khzAudio How can I solve the problem ? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: call load balancing
However: this means that a bug that makes Asterisk consume 100%CPU will practically stall the system. The system will still answer pings and open sockets, because that is done in the kernel. But nothing further will be done. Correct. Same if, for any reason, you have a loop in your dial-plan. Box is dead but responds to pings, etc. Not exactly a great solution killing everything on a server because of one buggy process. That is why I don't use -p. Can anyone comment on the performance differences with the 2.4 vs. 2.6 kernel on a semi-loaded (meaning not idle) box? I seem to get better responsiveness on a 2.6 kernel. snip htb on Linux works fine for me on the same box for outgoing traffic shaping. Can be the same machine that is running Asterisk as long as it's also acting as a gateway and nothing else competes for the link. Seems to improve flow quite a bit on a congested link. snip As far as concurrent calls, on 6 Mbit down / 768 kbit up (actual TCP throughput) cable link we can get 10 concurrent calls in G711, via IAX trunked with no quality loss. Jitter, if anything, seems to be a problem. At peak times it's about 40 ms. Dejittering at the other end is pretty much a must. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue
Mark Johnson schrieb: Jason wrote: Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then asterisk segfaults. Man... Use chan_sccp from Sergio at: ftp://ftp.berlios.de/pub/chan-sccp/ He is the most helpful person I've ever met. If you find a bug, report it to him, and it's usually fixed by the next day!! I don't have the same phone, but I've used 7910/40/60 with sccp and it works! ...and he is on vacation at the moment to recover from all the hot bug fixing :-) Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incompatible destination (88) Error Message. Please Help !!!
I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial(SIP/IZ-bc0a, Zap/g1/3118) in new stack Called g1/3118 Channel 0/1, span 1 got hangup Hungup 'Zap/1-1' No one is available to answer at this time I have tried to use PRI debug span 1 option and I am getting a strange error message Incompatible destination (88) Error Message. I am also attaching the complete degub information : -- Executing Dial(SIP/IZ-2f18, Zap/g1/3118) in new stack -- Making new call for cr 32773 Protocol Discriminator: Q.931 (8) len=36 Call Ref: len= 2 (reference 5/0x5) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0Location: User (0) Ext: 1 Progress Description: Calling equipmentis non-ISDN. (3) ] [28 03 b1 49 5a] Display (len= 3) Charset: 31 [ IZ ] [6c 03 21 80 33] Calling Number (len= 5) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3' ] [70 05 80 33 31 31 38] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '3118' ] -- Called g1/3118 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32773/0x8005) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 d8] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Incompatible destination (88), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == No one is available to answer at this time Any Ideas Please . I am realy stack here ! Thanks Iraklis Zografos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disable Call Waiting On SIP User Agents
Hi how to disable call waiting on SIP User agents (incominglimit=1 is Deprecated , End of life already announced no idea how to use setgroup to achieve same functionality) Thanks Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: call load balancing
On Sat, Aug 13, 2005 at 12:41:15AM -0700, Luki wrote: However: this means that a bug that makes Asterisk consume 100%CPU will practically stall the system. The system will still answer pings and open sockets, because that is done in the kernel. But nothing further will be done. Correct. Same if, for any reason, you have a loop in your dial-plan. Box is dead but responds to pings, etc. Not exactly a great solution killing everything on a server because of one buggy process. That is why I don't use -p. http://lists.digium.com/pipermail/asterisk-dev/2005-May/012434.html http://lists.digium.com/pipermail/asterisk-dev/2005-May/012439.html Can anyone comment on the performance differences with the 2.4 vs. 2.6 kernel on a semi-loaded (meaning not idle) box? I seem to get better responsiveness on a 2.6 kernel. You should expect that. Look for more information about preemption. This is one of the major changes between 2.4 and 2.6. htb on Linux works fine for me on the same box for outgoing traffic shaping. Can be the same machine that is running Asterisk as long as it's also acting as a gateway and nothing else competes for the link. Seems to improve flow quite a bit on a congested link. Asterisk wants fast access to the CPU. iptables and iproute's tc as well(?) allow you to define a more complex path in which each packet travels. This means: 1. more CPU time of the asterisk box invested in each packet. That task has a higher priority that Asterisk (even with -p) 2. delays on packets. This only really matters for packets of audio data. If you know what you're doing, then you actually don't delay (when unless really necessary) sound packets, so it should be fine. As for (1): I don't know tc. I've seen some ill-composed iptables rules that moves all packets through a complex set of rules. Usually one indication to that would be that the CPU spends non-negligable time in system. But if you have a zaptel card, this is something you should expect anyway. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Receive fax then send onwards
Hi,
I have managed to successfully receive faxes from a fax machine
attached to a Linksys PAP2, and send those faxes off via email using
rxfax (spandsp). From within the same process, I would now like to
automatically send the tiff file onwards as a fax using txfax out via
a zaptel interface. I have pasted my first cut at this below.
The problem here is that the SIP connection context on which I
received the fax seems to be where txfax wants to send the fax. I
imagine that I am missing something pretty fundamental in my
understanding of asterisk.
How do I create a new channel with the zaptel interface such that the
send will work?
Thanks,
bradley
[fax]
exten = _,1,Answer
exten = _,2,NoOp(${EXTEN})
exten = _,3,SetVar(DESTINATION=${EXTEN})
exten = _,4,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = _,5,rxfax(${FAXFILE})
exten = h,1,system(/var/lib/asterisk/scripts/mailfax ${FAXFILE}
[EMAIL PROTECTED] ${DESTINATION} )
exten = h,2,Goto(fax-send,${DESTINATION},1)
[fax-send]
;exten = _,1,Dial(${TRUNK}/${EXTEN},30)
exten = _,2,NoOp(${EXTEN})
exten = _,1,txfax(${FAXFILE}|caller|debug)
exten = _,2,Hangup()
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[Asterisk-Users] MISDN callerid
Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Identify call flow from manager events
Hi again, next problem I have is: I want to write an application which connects via manager api and displays the current telephone state. I know I have the action id to identify events which belong together. But if I have a call going inside asterisk and asterisk rings a phone these are two channels with different action ids. How can I know that these channels belong together? I know there are link events but what if the phone doesn't answer? Then I have two separate channels. Can I rely on the number in the action id after the dot? Eg. 111.0 is the incoming channel and 111.1 is the next channel belonging to the incoming call action I mean if the next channel which starts with 111. is created, can I assume that this channel belongs to the incoming channel? Background: My other problem was that MISDN displays the wrong callerid. I want to check if an incoming call is via MISDN and display the fixed callerid in my application (eg. a missing incoming call without a link event). Thanks Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call flow from manager events
On Sat, 2005-08-13 at 11:59 +0200, Christian Peter wrote: I know I have the action id to identify events which belong together. But if I have a call going inside asterisk and asterisk rings a phone these are two channels with different action ids. How can I know that these channels belong together? I know there are link events but what if the phone doesn't answer? Then I have two separate channels. With Asterisk stable all you get are the link events when the channels are linked. With Asterisk CVS-HEAD there is a dial event that informs your Manager application at dial time about the caller ids and the ids of the channels. Can I rely on the number in the action id after the dot? Eg. 111.0 is the incoming channel and 111.1 is the next channel belonging to the incoming call action not if other channels are created around that time. =Stefan signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disable Call Waiting On SIP User Agents
Gulzar Hussain wrote: Hi how to disable call waiting on SIP User agents Configure it on the SIP user agent! (incominglimit=1 is Deprecated , End of life already announced no idea how to use setgroup to achieve same functionality) We will have to change that. Incominglimit has an important role in the presence solution coming up for the SIP channel. The recommended way is to use the groupcount set of dialplan functions. Outgoinglimit will be removed, since it is disabled in the source code and has no function at all. Incominglimit works both ways now, so a change to call-limit would be better. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax on strike while rxfax works flawlessly
Hello, I have installed SpanDSP and the apps txfax an rxfax. Unfortunately I am having problems sending faxes. I only get cancelled transfers. I am trying to send a fax to a ISDN card connected to a zap channel. I am using following call file: --- Channel: ZAP/g1/25 MaxRetries: 0 WaitTime: 20 Application: txfax Data: /tmp/lVD3wy.tiff|caller Asterisk shows: --- -- Attempting call on ZAP/g1/25 for application txfax(/tmp/EmailToFax-lVD3wy.tiff|caller) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH == Primary D-Channel on span 2 up for TEI 64 Channel Zap/2-1 was answered. Lauching txfax(/tmp/lVD3wy.tiff|caller) on Zap/2-1 Urgent handler DIS with final frame tag In state 10 Start tx document CFR with final frame tag In state 4 Start tx page 0 Start tx page 1 XCN with final frame tag In state 14 -- Hungup 'Zap/2-1' Aug 13 13:43:55 NOTICE[11332]: pbx_spool.c:239 attempt_thread: Call completed to ZAP/g1/25 The tiff file I am using I have received and saved with rxfax... So I am reasonably sure that it has nothing to do with the image file. Unfortunatly the messages txfax prints do not tell me anything at all. Can anybody help me with this? Cheers, Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Fax
Hi, I plan on setting up an asterisk server to be used as an email-2-fax/fax-2-email server, for a company that sends and receives faxes almost 24/7 (milions of fax pages every month). From your experience in this, can Asterisk handle the heavy load? I intend to purchase a Saphir V PRI ISDN Adaptor for them. Thanks, Alex. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Push to talk and asterisk
Hi We are putting some efforts on having asterisk work as a PTT server over GPRS. Anyone interested to part of it , Please email me privately Best Regards Mustafa N. Deeb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash over SIP Trunk
Hello, I have a Incoming/Outgoing SIP Trunk setup to Broadvoice, is there a way to send a "Flash" over the trunk, for example, to do flash transfers and call-waiting? I tried to use Flash() but it seems to not work on the sip trunk, only my zap trunks. Please let me know, thanks! :) Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remotely rebooting Sipura SPA-3000 from command line
On Saturday 13 Aug 2005 07:29, Eric Bishop wrote: Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:[EMAIL PROTECTED]/admin/reboot The strange thing is it works fine when I go to http://admin:[EMAIL PROTECTED]/admin/reboot with my web browser... wget has specific commandline options for usernames and passwords. The admin:mypassword@ is stripped out by your brower but not in wget. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
On 11:44, Sat 13 Aug 05, Christian Peter wrote: Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian Hi, I think this is normal behavior for ISDN lines. My guess it's the telco not sending the 0. I have the same thing on AVM Fritz with chan_capi and with the QuadBRI with qozap and chan_zap. I simply add the 0 to national numbers (based on length of numbers without leading 0) or a + to international numbers. This fixed all later lookup functionality in my dialplan. Hope this is of any help for you. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
Michiel van Baak schrieb: I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. I think this is normal behavior for ISDN lines. My guess it's the telco not sending the 0. The callerid is tranmitted completely different. Adding zeros or plusses is a matter of the equipment (phone). I have the same thing on AVM Fritz with chan_capi and with the QuadBRI with qozap and chan_zap. I simply add the 0 to national numbers (based on length of numbers without leading 0) or a + to international numbers. This fixed all later lookup functionality in my dialplan. For chan_capi - capi.conf: [general] nationalprefix=0 internationalprefix=00 For HFC-cards (bristuff / ZAPHFC) - zapata.conf: [channels] pridialplan=local prilocaldialplan=local For MISDN: no idea... look into the docs or use ZAPHFC :-). Changing the CID in the dialplan is really ugly :-). Besides, Michiel, I strongly recommend chan_capi-cm! It has more functions than chan_capi. You can e.g. send any hangupcause you want back to the network e.g. 021 call rejected :-) ... chan_capi-cm - http://sourceforge.net/projects/chan-capi/ cause codes - http://www.telos-systems.com/?/techtalk/cause.htm -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
This chan_misdn version is old, use a newer one. It seems that TypeOfNumber interpretation has not been integrated in this verison. Best regards Hans Christian Peter schrieb: Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One more newbie question
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
On 14:43, Sat 13 Aug 05, Stefan Gofferje wrote: Michiel van Baak schrieb: I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. I think this is normal behavior for ISDN lines. My guess it's the telco not sending the 0. The callerid is tranmitted completely different. Adding zeros or plusses is a matter of the equipment (phone). I have the same thing on AVM Fritz with chan_capi and with the QuadBRI with qozap and chan_zap. I simply add the 0 to national numbers (based on length of numbers without leading 0) or a + to international numbers. This fixed all later lookup functionality in my dialplan. For chan_capi - capi.conf: [general] nationalprefix=0 internationalprefix=00 For HFC-cards (bristuff / ZAPHFC) - zapata.conf: [channels] pridialplan=local prilocaldialplan=local For MISDN: no idea... look into the docs or use ZAPHFC :-). Changing the CID in the dialplan is really ugly :-). Besides, Michiel, I strongly recommend chan_capi-cm! It has more functions than chan_capi. You can e.g. send any hangupcause you want back to the network e.g. 021 call rejected :-) ... chan_capi-cm - http://sourceforge.net/projects/chan-capi/ cause codes - http://www.telos-systems.com/?/techtalk/cause.htm Stefan, The nationalprefix settinf fixed it, thnx. The qozap cards uses a patched fixcid.agi, taken from AAH. Works great for the job it does at customers place. I'm already testing with chan_capi-cm, but for some reason the capiAnswerFax never generates a faxfile with more than 0 bytes in it. Because I only have ISDN at work I didn't yet have time to really dive into it and reverted to 0.3.5 for the moment. Maybe later today I have some time to test, it is weekend afterall ;) Thanks for the link of the cause codes, I can already think of some nice implementations here. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] receiving calls from FWD
I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] . register = 688426:[EMAIL PROTECTED]/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426 secret=xxx dtmfmode=inband host=fwd.pulver.com port=5060 nat=yes canreinvite=no externip=194.185.53.47 localnet=192.168.1.0/255.255.255.0 I orignally tried without the last two lines, then added these and forwarded all ports from the NAT to the asterisk server, but still no result. Is anyone with asterisk behind a NAT successfully receiving calls from FWD? Can you give me any pointers on the above configuration? I realize that I could also try IAX which is supported by FWD, but I'm having this problem with another SIP provider too, so I'd like to get it working for SIP. Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] receiving calls from FWD
On 15:14, Sat 13 Aug 05, John Fawcett wrote: I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] . register = 688426:[EMAIL PROTECTED]/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426 secret=xxx dtmfmode=inband host=fwd.pulver.com port=5060 nat=yes canreinvite=no externip=194.185.53.47 localnet=192.168.1.0/255.255.255.0 I orignally tried without the last two lines, then added these and forwarded all ports from the NAT to the asterisk server, but still no result. Is anyone with asterisk behind a NAT successfully receiving calls from FWD? Can you give me any pointers on the above configuration? I realize that I could also try IAX which is supported by FWD, but I'm having this problem with another SIP provider too, so I'd like to get it working for SIP. Hi, Those 2 lines about externip and localnet should go before any register or phone/provider stanza. Check if you are registered to fwd/other sip provider. In asterisk CLI type: sip show registry. If you forwarded all ports that are stated in rtp.conf and the 5060 to the asterisk box it should work. At least it does here. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
Tom Rymes wrote: Chris, Maybe you could write a generic config file and post it to the wiki? I tried to post as a comment but the XML was excluded. How do I do that? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Help, using SendText cmd sip message...
Any idea, how to send sip messages in a call to any of then if they are sip or just one is a sip device, i check the code in sendtext but i dont know how to change the current channel i need to send the message to the other side, how can i know to whom is connected a channel and change the code to obtain that information... -- Forwarded message -- From: Roddy G. Posada Santos [EMAIL PROTECTED] Date: 11-ago-2005 0:18 Subject: Help, using SendText cmd sip message... To: Asterisk-Users@lists.digium.com i have a sip phone that handle SIP MESSAGE METHOD, i need that asterisk send a message to this phone when it makes a call, but the sendText cmd send the message in the other direction, do you know how to change this behavior that asterisk send the sip message to the caller, not to calle sip message from asterisk to caller: SIPPHONE--SIPMESSAGE---ASTERISKANYCHANNEL SIP PHONE - ASTERISK --ANY CHANNEL(SIP/ZAP/IAX) CALLER---CALLEE SendText cmd only do in this way: CALLER--SIPMESSAGE---CALLEE -- -- roddy -- Roddy G. Posada Santos Ofic. 593-2-2438308 Cel. 593-9-9719212 -- -- roddy -- Roddy G. Posada Santos Ofic. 593-2-2438308 Cel. 593-9-9719212 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
ISDN Setup [was: Re: [Asterisk-Users] One more newbie question]
[ Subject changed so people looking at the list index will actually have the minimal clue as to what this post is about ]. On Sat, Aug 13, 2005 at 01:50:16PM +0100, Sean Rima wrote: Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. And obviously you want us to do all of your work for you? You don't even give a meaningful subject to your messages, so in the future the discussions that followed will be useless to searchers. I didn't even bother reading previous newbie question threads. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? No. Does that answer your question? Now go and do the minimal search: * What type of ISDN services is availble at your country/area? * Did you read a bit about the availble ISDN support in Asterisk? - It is generally better to ask questions that indicate you did and point to parts you don't understand, than just ask general qustions. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Fax
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandru Thomae Sent: Saturday, August 13, 2005 7:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Fax Hi, I plan on setting up an asterisk server to be used as an email-2-fax/fax-2-email server, for a company that sends and receives faxes almost 24/7 (milions of fax pages every month). From your experience in this, can Asterisk handle the heavy load? I intend to purchase a Saphir V PRI ISDN Adaptor for them. Thanks, Alex. Alex, Skip asterisk if your goal is high-volume faxing. HylaFAX is a much better suited product. www.hylafax.org Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One more newbie question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima Sent: Saturday, August 13, 2005 8:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] One more newbie question Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? Sean This is not that difficult. The first question you need to answer is whether you want to use a dedicated circuit (E1/T1/PRI) or multiple copper lines. This mostly depends on the call volume that you will be handling. Depending on your choice, you can install either a Sangoma or a Digium card to handle a dedicated circuit or you can use a Digium TDM400B card or a PSTN gateway to connect to your POTS phone lines. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Push to talk and asterisk
You may be duplicating work that has already been done. http://www.zapatatelephony.org/app_rpt_article.pdf Mark, KC2ENI Mustafa N. Deeb wrote: Hi We are putting some efforts on having asterisk work as a PTT server over GPRS. Anyone interested to part of it , Please email me privately Best Regards Mustafa N. Deeb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue turned to chan_sccp issue.
Couple questions since I finally got chan_sccp to work. 1. Has anyone successfully gotten it to work behind NAT, i called my isp and asked them for 35 more ip's and they laughed at me + the cost of getting them would far outdo the benifits. and 2. can someone show me a config with a multi phone configuration Jason Stefan Gofferje wrote: Mark Johnson schrieb: Jason wrote: Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then asterisk segfaults. Man... Use chan_sccp from Sergio at: ftp://ftp.berlios.de/pub/chan-sccp/ He is the most helpful person I've ever met. If you find a bug, report it to him, and it's usually fixed by the next day!! I don't have the same phone, but I've used 7910/40/60 with sccp and it works! ...and he is on vacation at the moment to recover from all the hot bug fixing :-) Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Card (Rev G) with bad FXS module?
I think I have a bad FXS module on my TDM400P. Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The TDM400P is rev G if that means anything. When I do a ls -l /proc/zaptel/* there's nothing there. I'm pretty sure my configs are correct. I copied them verbatim from http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html and my card is setup the same way with the green FXS Module in TEL1 and the red FXO module in TEL2. (I believe thats the standard way they come from Digium.) I purchased this probably two years ago. I've tried both the stable zaptel and CVS zaptel. I tried moving the modules to 3 and 4 (and changed everything accordingly). I even tried this: Comment out this line on zaptel/zconfig.h and recompile zaptel. /* * Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's */ /* #define TDM_REVH_MATCHALL */ This is just a basic test setup for my home. I want to get an analog phone and 2 Cisco 7910 phones working. Any help would be appreciated. Jeff Borders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ISDN Setup [was: Re: [Asterisk-Users] One more newbie question]
Tzafrir Cohen wrote: [ Subject changed so people looking at the list index will actually have the minimal clue as to what this post is about ]. On Sat, Aug 13, 2005 at 01:50:16PM +0100, Sean Rima wrote: Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. And obviously you want us to do all of your work for you? Nope, I admit that I should have done a better subject line You don't even give a meaningful subject to your messages, so in the future the discussions that followed will be useless to searchers. I didn't even bother reading previous newbie question threads. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? No. I read an online doc and it was a great help, shold ahve searched before I posted the message Does that answer your question? Now go and do the minimal search: * What type of ISDN services is availble at your country/area? * Did you read a bit about the availble ISDN support in Asterisk? - It is generally better to ask questions that indicate you did and point to parts you don't understand, than just ask general qustions. Yeah true and I will bear this in mind in the future Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One more newbie question
Tom Rymes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima Sent: Saturday, August 13, 2005 8:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] One more newbie question Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup or not? Sean This is not that difficult. The first question you need to answer is whether you want to use a dedicated circuit (E1/T1/PRI) or multiple copper lines. This mostly depends on the call volume that you will be handling. Depending on your choice, you can install either a Sangoma or a Digium card to handle a dedicated circuit or you can use a Digium TDM400B card or a PSTN gateway to connect to your POTS phone lines. For my own needswhich is the primary one, it will just be connected to one channel of the ISDN circuit, the other channel is taken by the PC connection to the net. I have seen a web site that deals with a lot of my questions so I will be reading it in some detail Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Beta IAX Statistics Program
Hot off the wire: http://www.sineapps.com/news.php?rssid=927 Hi, we have put together a small application for Windows to allow you to check IAX network statistics. Basically all you need is the .Net framework and the user/pass/host/extension/context details. There is one parameter available when you start. This is dial string. It is made up as follows: user:[EMAIL PROTECTED]/[EMAIL PROTECTED] You can download it from here: (415K) http://www.sineapps.com/SineStatsIAX%20Installer.zip Please let me know how you go with it. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Card (Rev G) with bad FXS module?
lspci -v what output do you get? Also, what OS are you using? Jeff Borders wrote: I think I have a bad FXS module on my TDM400P. Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The TDM400P is rev G if that means anything. When I do a ls -l /proc/zaptel/* there's nothing there. I'm pretty sure my configs are correct. I copied them verbatim from http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html and my card is setup the same way with the green FXS Module in TEL1 and the red FXO module in TEL2. (I believe thats the standard way they come from Digium.) I purchased this probably two years ago. I've tried both the stable zaptel and CVS zaptel. I tried moving the modules to 3 and 4 (and changed everything accordingly). I even tried this: Comment out this line on zaptel/zconfig.h and recompile zaptel. /* * Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's */ /* #define TDM_REVH_MATCHALL */ This is just a basic test setup for my home. I want to get an analog phone and 2 Cisco 7910 phones working. Any help would be appreciated. Jeff Borders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P Card (Rev G) with bad FXS module?
Jeff: Which operating system are you running? From: Jeff Borders [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDM400P Card (Rev G) with bad FXS module? Date: Sat, 13 Aug 2005 10:46:20 -0400 I think I have a bad FXS module on my TDM400P. Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The TDM400P is rev G if that means anything. When I do a ls -l /proc/zaptel/* there's nothing there. I'm pretty sure my configs are correct. I copied them verbatim from http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html and my card is setup the same way with the green FXS Module in TEL1 and the red FXO module in TEL2. (I believe thats the standard way they come from Digium.) I purchased this probably two years ago. I've tried both the stable zaptel and CVS zaptel. I tried moving the modules to 3 and 4 (and changed everything accordingly). I even tried this: Comment out this line on zaptel/zconfig.h and recompile zaptel. /* * Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's */ /* #define TDM_REVH_MATCHALL */ This is just a basic test setup for my home. I want to get an analog phone and 2 Cisco 7910 phones working. Any help would be appreciated. Jeff Borders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Phone- 7905G
Has anybody used a Cisco 7905G or similar model with Asterisk using skinny? How can i set it up with an asterisk box? Thanks, Orlando ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] premature call release - SIP 480
When executing: Dial (SIP/[EMAIL PROTECTED],60) I get about 15 seconds of ringing, the called party rings, but if not answered in the ~15 seconds I get back SIP 480 temporarily unavailable. If the call is answered everything is fine and the call will continue as expected. The call is being passed to a TNT media gateway then to the PSTN via a PRI The TNT reports Q850 cause 19 and responds with SIP 480 Somehow the TNT thinks the called stopped progressing on the PRI after 15 to 20 seconds. The Telco says they have done a capture and are getting a normal release, in other words their switch is not terminating the call or sending any Q850 message. I can not find any timers in the TNT that might cause this, and it is not reporting any expired timers. Any ideas? Does the SIP INVITE from * to the TNT contain a timeout? If so is it possible the, 60 in the dial command is being ignored? Either; The TNT got a maximum time parameter from asterisk and it has been exceeded, so the TNT responds 480, or; The TNT has a timer that expires after n seconds and sends the 480 on its own, or; The Telco is not seeing the progress they want to see and is sending the Q850 cause 19. Any opinions, suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue turned to chan_sccp issue.
Jason wrote: Couple questions since I finally got chan_sccp to work. 1. Has anyone successfully gotten it to work behind NAT, i called my isp and asked them for 35 more ip's and they laughed at me + the cost of getting them would far outdo the benifits. and 2. can someone show me a config with a multi phone configuration I don't know about the nat, but don't think it should be a problem. It does not work like SIP where the port stuff is determined via the sip protocl. In your sccp.conf file, you have the [devices] There you enter the device settings and then the device = SEPxxx And add some more settings and then your device= SEPxxx and so on for as many devices as you need. You need to then go to the [lines] and add at least one line for each device as well. Are you using the 20050805 release? You can look at the sample sccp.conf file for the settings. ftp://ftp.berlios.de/pub/chan-sccp/ There is a mailing list there as well. He is the most helpful person I've ever met. If you find a bug, report it to him, and it's usually fixed by the next day!! I don't have the same phone, but I've used 7910/40/60 with sccp and it works! I have to agree, he has been very helpfull. ...and he is on vacation at the moment to recover from all the hot bug fixing :-) -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone- 7905G
Orlando Guitián wrote: Has anybody used a Cisco 7905G or similar model with Asterisk using skinny? How can i set it up with an asterisk box? Are you using the latest version of chan_sccp? http://www.voip-info.org/tiki-index.php?page=chan_sccp2 The driver link can be gotten directly from there. There is a mailing list just for chan_sccp. Try the latest version and let us know what happens when you try to register. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions exchange
Hi all, I just connected 4 * box (by IAX) and now i'm thinking about this: can i exchange the extensions list between this boxs ? The clinets/phones can known which other clients are connected ? Thanks, Gio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended Trasnfer
Hi, I'm having prolems with attended transfer configuration. Does this feature had been implemmented by any of you??? What is the best * version to make this work?? Some sample example?? I'm using like this in features.conf: [featuremap] atxfer = 900 ; Attended transfer My * version was intalled by [EMAIL PROTECTED] CD and is 1.0.9. Thanks in advance. Jônatas Amorim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
My linux box speaks pppoe to external DSL modem. Nortel NTEX35 BAAB. It's up 24/7 and provides web service...etc. Has 6 nics, one of them is fiber. Asterisk is on the same box. Don't have any IP phones yet. The asterisk default is to listen on all 6 enet interfaces? (this is what I'd want). Etherape is showing me a healthy dose of subnet broadcast on each nic...which I assume is asterisk doing discovery, or it's announcing itself? asteriskdocs.org says this: We now having a working example of a zapata.conf with an FXS channel (1) and an FXO channel (4) that we can use. Channel 4 can be connected to an analog circuit such as might be provided by your phone company. You can plug an analog telephone directly into Channel 1. The digium board will be in the same box. Does this mean: Channel 4 to incoming phone line. Channel 1 to DSL modem? Or DSL modem to the incoming line...and then the pass thru port on the DSL modem goes to Channel 4? Will this even work? I'd hate to have to switch to a cable modem. TIA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 86
Hello All i need to transfer CDR data from linux to MS SQL Serever (on Windows). writing by Perl. I have download and install UnixODBC, DBI, DBD from CPAN, when i tested isql -DSN -UID - PWD, that's successful, but when run by perl, message alert could not loaded driver database, anybody used ODBC for linux, connect to MS SQL Sevre, please help me? easysoft and openlink have trial license, run with iODBC, run on 30 days, could you like help me don't used iODBC, free tools. any advice? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 decoding
Hi, I searched a while about T.38 decoding, and learned about the bounty for T.38 support for asterisk and some softdecoders and some hardware de- and encoding T.38. Now I wonder, if there is already any (almost) ready to use solution for decoding of T.38 faxes? My szenario would be: - Receiving a SIP call (containing the T.38 fax) by my provider with my asterisk box. - asterisk would forward that SIP call to the converter. - The converter would send the SIP call back to my asterisk box, but now with the fax deocoded to an ordenary anolog fax. Has anyone experience with a working solution, maybe a foreign service provider doing it, or a working (asterisk independent) software? Thanks for any hints! Roger. P.S. Currently I'm trying to understand, what ionidea's T.38 software is already able to do, but I'm still confused. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Sat, Aug 13, 2005 at 08:10:03AM -0800, Cliff Savage wrote: The digium board will be in the same box. Does this mean: Channel 4 to incoming phone line. Channel 1 to DSL modem? Or DSL modem to the incoming line...and then the pass thru port on the DSL modem goes to Channel 4? Will this even work? I'd hate to have to switch to a cable modem. ADSL should not bother PSTN as long as you use a proper filter. In our case a proper filter was supplied by the phone company when we installed the ADSL line. We Happily use Asterisk with an FXO card and an ADSL connection from the same phone line. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Fax
I plan on setting up an asterisk server to be used as an email-2-fax/fax-2-email server, for a company that sends and receives faxes almost 24/7 (milions of fax pages every month). From your experience in this, can Asterisk handle the heavy load? I intend to purchase a Saphir V PRI ISDN Adaptor for them. Thanks, Alex. Alex, Skip asterisk if your goal is high-volume faxing. HylaFAX is a much better suited product. www.hylafax.org Tom I'll second that. Hylafax has can handle the job. If you put asterisk in between you are looking for problems. I've the following setup working with asterisk NVBackgroundDetect implemented. PSTN -- asterisk -- hylafax It woks, I would say 90% of the time. There seems to me some timing problems with asterisk, see my posting with subject: real-time priority -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Push to talk and asterisk
Has anyone been able to compile app_rpt? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Saturday, August 13, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Push to talk and asterisk You may be duplicating work that has already been done. http://www.zapatatelephony.org/app_rpt_article.pdf Mark, KC2ENI Mustafa N. Deeb wrote: Hi We are putting some efforts on having asterisk work as a PTT server over GPRS. Anyone interested to part of it , Please email me privately Best Regards Mustafa N. Deeb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Load Testing
How do you generate those calls? That's what Im interested about.. I do have multiple asterisk servers that I can use to send the calls but how to generate them.. That's the question. :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tim Connolly |Sent: Viernes, 12 de Agosto de 2005 09:00 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Load Testing | |I could probably shoot about 115 calls towards you, would that do ? | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Friday, August 12, 2005 8:56 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Load Testing | |Guys. | |How and which tools to use to load test an asterisk install? |Say for example, you need to see how many calls can be routed |thru before losing quality and making the cpu jump to the roof? | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Load Testing
Hi Michael. Are there any script already made for doing this? Sending calls from one asterisk to the one been tested? Something that would simulate your 1 phone scenario? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |The VoIP Connection |Sent: Viernes, 12 de Agosto de 2005 10:42 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Load Testing | |Anton, | |A great tool for ghetto call capacity testing is a single snom phone. |There is no limit to how many calls a snom phone can make, |just put it on hold and dial again. So, with a single snom |phone and a little imagination you can test any number of |scenarios. You can approximate basic SIP capacity by creating |an extension that plays the asterisk test message and dialing |it repeatedly until quality starts to degrade or asterisk gives up. |To simulate actual call throughput you really need another |(faster) machine to connect to, but you can use the same technique. | |You can run top on the console while you are doing your |tests to see what resources you are using. Check your logs |when you are done to see what errors were generated when it |came unglued. CPU is not always the limiting resource, |especially with Digium card interfaces which tend to be bound |by FSB speed, but echo cancellation and codec conversion will |burn a LOT of cycles. | |Michael Crown |Managing Partner |www.thevoipconnection.com |321.989.6728 ext. 611 |sip:[EMAIL PROTECTED] | | -Original Message- | From: Anton Krall [mailto:[EMAIL PROTECTED] | Sent: Friday, August 12, 2005 9:56 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: [Asterisk-Users] Load Testing | | Guys. | | How and which tools to use to load test an asterisk install? | Say for example, you need to see how many calls can be routed thru | before losing quality and making the cpu jump to the roof? | | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Testing
I just create a bunch of call files and copy (Yes I know you are supposed to move them) into the outgoing calls directory. Darren Wiebe [EMAIL PROTECTED] Anton Krall wrote: How do you generate those calls? That's what Im interested about.. I do have multiple asterisk servers that I can use to send the calls but how to generate them.. That's the question. :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tim Connolly |Sent: Viernes, 12 de Agosto de 2005 09:00 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Load Testing | |I could probably shoot about 115 calls towards you, would that do ? | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Friday, August 12, 2005 8:56 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Load Testing | |Guys. | |How and which tools to use to load test an asterisk install? |Say for example, you need to see how many calls can be routed |thru before losing quality and making the cpu jump to the roof? | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggestions for mainstream hardware compatible with TE411P.
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote:    I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or … any other brand and model that is known to work well with the TE411P ? Will an old Proliant do?I've built five PBXes on Dell Dimension 2600s that run flawlessly. They're P3 2.6GHz machines, so processor load stays super-low. Using a combination of TE110Ps and VoIP termination/origination, across ~35 users at each location on 7960s. Never missed a beat.I would consider a "consumer" box with a strong CPU over an old server, then spend your money on an ATA RAID card and mirror everything for disaster recovery.Hope that helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Load Testing
Try this: phone1=192.168.7.251 number1=1+0+1 curl http://$phone1/command.htm?key=$number1+ENTER; /dev/null 2/dev/null sleep 10 curl http://$phone2/command.htm?key=CANCEL; /dev/null 2/dev/null Available keys: #define KEY_CANCEL CANCEL #define KEY_CLEAR CLEAR #define KEY_ENTER ENTER #define KEY_OFFHOOK OFFHOOK #define KEY_ONHOOK ONHOOK #define KEY_RIGHT RIGHT #define KEY_LEFT LEFT #define KEY_FUNCTION FUNCTION // below redial #define KEY_MENU MENU #define KEY_REDIAL REDIAL #define KEY_ORG_F1 F1 #define KEY_ORG_F2 F2 #define KEY_ORG_F3 F3 #define KEY_ORG_F4 F4 #define KEY_SPEAKER SPEAKER #define KEY_DISCONN DISCONNECT #define KEY_RECALL RECALL #define KEY_BREAK BREAK #define KEY_TRANSFER TRANSFER #define KEY_CONFERENCE CONFERENCE #define KEY_HELP HELP #define KEY_VOLUME_UP VOLUME_UP #define KEY_VOLUME_DOWN VOLUME_DOWN #define KEY_MUTE MUTE #define KEY_HEADSET HEADSET #define KEY_UP UP #define KEY_DOWN DOWN #define KEY_REC REC #define KEY_RETRIEVE RETRIEVE #define KEY_SETTINGS SETTINGS #define KEY_PHONE_BOOK PHONE_BOOK #define KEY_SNOM SNOM #define KEY_DND DND And yes, it is a good reason to set the password on your phone if you dont want to use it only for testing! Giving everyone access to the web server of the phone is not a good idea, not only for snom phones. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, August 13, 2005 7:46 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Load Testing Hi Michael. Are there any script already made for doing this? Sending calls from one asterisk to the one been tested? Something that would simulate your 1 phone scenario? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of The VoIP |Connection |Sent: Viernes, 12 de Agosto de 2005 10:42 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Load Testing | |Anton, | |A great tool for ghetto call capacity testing is a single snom phone. |There is no limit to how many calls a snom phone can make, just put it |on hold and dial again. So, with a single snom phone and a little |imagination you can test any number of scenarios. You can approximate |basic SIP capacity by creating an extension that plays the asterisk |test message and dialing it repeatedly until quality starts to degrade |or asterisk gives up. |To simulate actual call throughput you really need another |(faster) machine to connect to, but you can use the same technique. | |You can run top on the console while you are doing your tests to see |what resources you are using. Check your logs when you are done to see |what errors were generated when it came unglued. CPU is not always the |limiting resource, especially with Digium card interfaces which tend to |be bound by FSB speed, but echo cancellation and codec conversion will |burn a LOT of cycles. | |Michael Crown |Managing Partner |www.thevoipconnection.com |321.989.6728 ext. 611 |sip:[EMAIL PROTECTED] | | -Original Message- | From: Anton Krall [mailto:[EMAIL PROTECTED] | Sent: Friday, August 12, 2005 9:56 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: [Asterisk-Users] Load Testing | | Guys. | | How and which tools to use to load test an asterisk install? | Say for example, you need to see how many calls can be routed thru | before losing quality and making the cpu jump to the roof? | | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Suggestions for mainstream hardware compatiblewith TE411P.
Thanks for the suggestion. One of my problems is that a TE110P worked flawlessly in my MPC server. As soon as I upgraded to the TE411P, I started having all sorts of issues. The biggest being an IRQ conflict, which was resolved but only to find I still get kernel panics under minor load. I think Im finding myself victim of early-adopter syndrome. I havent been able to get much feedback (no pun intended) from owners of the TE411Ps. Anyone want to trade a TE411P for 4 TE110Ps ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Saturday, August 13, 2005 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Suggestions for mainstream hardware compatiblewith TE411P. On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or any other brand and model that is known to work well with the TE411P ? Will an old Proliant do? I've built five PBXes on Dell Dimension 2600s that run flawlessly. They're P3 2.6GHz machines, so processor load stays super-low. Using a combination of TE110Ps and VoIP termination/origination, across ~35 users at each location on 7960s. Never missed a beat. I would consider a consumer box with a strong CPU over an old server, then spend your money on an ATA RAID card and mirror everything for disaster recovery. Hope that helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] generic x100p always OnHook (FXO port trhoug optimum voice VOIP service)
Hi guys, First, changed the thread to generic x100p always OnHook since I realized my problem is more general than to just optimum voice. My x100p card is always onhook, and that's why it kills the dialtone whenever I connect it to the phone line. Some more in depth debug/status: zap show channel 1: InAlarm: 1 (believe correspond to red alarm) ... Actual Hookstate: OnHook Ichanged the rx/tx but I don't believe this will fix the problem. Also tried with loop/ground start signaling but no luck. help please... carlos --- Madhawa Jayanath [EMAIL PROTECTED] wrote: Carlos Trallero wrote: Hello, I have asterisk running on Fedora Core 3 with a x100p (oem). After some time I got asterisk with some soft extensions working (u gotta love open source), but I'm stuck with outbound dialing. This is the diagnose: - detect 1 wcfxo channel. - when trying to make an outside call I get unable to create channel of type Zap. Everyone is busy/congested at this time - When I plug the x100p to the phone jack, the dial tone in all of my phones die. Because of the later I'm suspecting that there is some problem with the signaling or voltage detection. My PSTN line is actually from a VoIP modem that runs over the Cablevision network (known as Optimum Voice). Thanks everyone. Carlos __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Where did u get that OEM X100P? Is it MD3200 chip? Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Card (Rev G) with bad FXS module?
On Sat, Aug 13, 2005 at 01:21:13PM +0700, Zvi Kushnaroff wrote: Jeff, You might want to try changig the setup of tzaptel.conf and zapata.conf. I have a TDM400P with 2 FX0 modules and one FXS module. I found that the stting that digium recommends are WRONG. It works fine with the following seetings: As U can see no channel 2!! zaptel.conf fxoks=1 fxsks=3,4 That depends on which modules were populated in the card. So it ca be any wild combination, basically. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call flow from manager events
Hi Stefan, thanks for the immediate response. Luckily I found a fix to my MISDN problem so I don't have to rely on the channel information. Thanks, Christian Am Samstag, den 13.08.2005, 10:53 + schrieb Stefan Reuter: On Sat, 2005-08-13 at 11:59 +0200, Christian Peter wrote: I know I have the action id to identify events which belong together. But if I have a call going inside asterisk and asterisk rings a phone these are two channels with different action ids. How can I know that these channels belong together? I know there are link events but what if the phone doesn't answer? Then I have two separate channels. With Asterisk stable all you get are the link events when the channels are linked. With Asterisk CVS-HEAD there is a dial event that informs your Manager application at dial time about the caller ids and the ids of the channels. Can I rely on the number in the action id after the dot? Eg. 111.0 is the incoming channel and 111.1 is the next channel belonging to the incoming call action not if other channels are created around that time. =Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
Hi Hans, it was there but it doesn't work in this particular version. It works with current daily snapshot. Thanks, Christian Am Samstag, den 13.08.2005, 14:47 +0200 schrieb Johann Steinwendtner: This chan_misdn version is old, use a newer one. It seems that TypeOfNumber interpretation has not been integrated in this verison. Best regards Hans Christian Peter schrieb: Hi all, I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. Is anyone seeing this behaviour too? Thanks in advance Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue turned to chan_sccp issue.
Jason schrieb: Couple questions since I finally got chan_sccp to work. 1. Has anyone successfully gotten it to work behind NAT, i called my isp and asked them for 35 more ip's and they laughed at me + the cost of getting them would far outdo the benifits. and 2. can someone show me a config with a multi phone configuration 1.) No NAT... SCCP is NOT nat-capable. But it works fine with VPN. You anyway don't want to talk plain SCCP/RTP via the internet without encryption. Real-time splitting and copying a RTP-stream is really easy going! 2.) Multi-phone sccp.conf or extensions.conf? Here's a sccp.conf: [general] keepalive = 30 ; IMPORTANT: 5secs. lead to trouble with ; 7960 context = internal dateFormat = D.M.Y ; M-D-Y in any order (5 chars max) bindaddr = 192.168.1.200 ; asterisk box. port = 2000 ; listen on port 2000 (Skinny, default) debug = 0 [devices] type= 7905 description = Bedroom tzoffset= 0 autologin = 6004 device = SEP type= 7960 description = Office tzoffset= 0 autologin = 6000 speeddial = 6001,6001,[EMAIL PROTECTED] speeddial = 6004,6004,[EMAIL PROTECTED] device = SEP type= 7960 description = LivingRoom tzoffset= 0 autologin = 6001 speeddial = 6000,6000,[EMAIL PROTECTED] speeddial = 6004,6004,[EMAIL PROTECTED] device = SEP [lines] id= 6000 pin = 1234 label = 6000 description = Office context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Office cid_num = 6000 line = 6000 id= 6001 pin = 1234 label = 6001 description = LivingRoom context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Living Room cis_num = 6001 line = 6001 id= 6004 pin = 1234 label = 6004 description = Bedroom context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Bedroom cid_num = 6004 line = 6004 Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] False Zap answer problem
Hi All, I am experiencing a very strange problem. I call the FXO channels (Zap/2 and 3) almost at the same time, and then hang both up. The operator extension is Zap/6, and after the greeting message Zap/6 starts to ring (there is no disconnect supervision here, and I disabled the busy detect for hangup detection, so the outside lines stay open until Asterisk hangs them up). As you can see on the following CLI section, when the VoiceMail promt starts to play for Zap/2, the system thinks that Zap/6-2 has answered the Zap/3, when in fact nobody answers it: -- Hungup 'Zap/6-1' -- Executing VoiceMail(Zap/2-1, u0) in new stack -- Playing 'vm-theperson' (language 'tr') -- Zap/6-2 answered Zap/3-1 I can very easily replicate this (did it 3 times). Where should I look for the source of this problem? Is it the TDM card or the Asterisk? How does Asterisk know if a line has been answered on a TDM card? This seems like some kind of cross-talk between the two FXO channels, but it's really strange. Has anybody had a similar problem? Of course, the workaround is to use some sort of hangup detection on outside lines, but I was having false hangup problems with busy detect, thus disabled it and noticed this problem (and see my above comment on disconnect supervision here). I have 2x TDM cards (version E/F) with 3x FXO and 4x FXS modules on each. Asterisk version is: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-10 22:55:45 UTC I have echocancel and echocancelwhenbridged enabled. (Could it be the echocanceller?) I would appreciate any help, Soner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Henning G. Schulzrinne quote on IAX2 from von magazine
[thread moved from -dev due to non-dev content] At 6:40 PM +0200 on 8/13/05, Andreas Sikkema wrote: On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote: He doesn't seem to really understand the strengths and weaknesses of IAX. IAX has drawbacks, but none of the problems he lists actually exist. OK, I'll bite ;-) How would IAX2 solve trombones? -- Andreas Sikkema Since this is a very vague accusation of protocol shortcoming, I'll answer in a very vague way: IAX2 has the ability to native bridge two endpoints together, even if the call was established by a third party. Media is not tromboned through the third party if it is possible for the two endpoints to communicate directly, and it is administratively permitted. Please read the (not quite completed) IAX2 specs. http://www.cornfed.com/iax.pdf JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 decoding
You do realize that t.38 is the act of taking the t.30 stream and stuffing into UDPTL packet and sending it over a network with a little ASN.1 header added and some reliable delivery kinda like how IAX has reliable delivery of UDP packets used for signaling. This is a very basic description of how its done Go check how t38modem does it.. it emulates a modem and just intercepts the t.30 stream and transports it. /b On Aug 13, 2005, at 11:55 AM, Roger Schreiter wrote: Hi, I searched a while about T.38 decoding, and learned about the bounty for T.38 support for asterisk and some softdecoders and some hardware de- and encoding T.38. Now I wonder, if there is already any (almost) ready to use solution for decoding of T.38 faxes? My szenario would be: - Receiving a SIP call (containing the T.38 fax) by my provider with my asterisk box. - asterisk would forward that SIP call to the converter. - The converter would send the SIP call back to my asterisk box, but now with the fax deocoded to an ordenary anolog fax. Has anyone experience with a working solution, maybe a foreign service provider doing it, or a working (asterisk independent) software? Thanks for any hints! Roger. P.S. Currently I'm trying to understand, what ionidea's T.38 software is already able to do, but I'm still confused. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:(2) Henning G. Schulzrinne quote on IAX2 from von magazine
[moved from -dev list due to non-dev topic content] At 12:44 PM +0800 on 8/13/05, Steve Underwood wrote: Mike Taht wrote: but hey, maybe the folk on this list understand where he's coming from and can explain why sip is better He is one of originators of RTP and the main guy behind SIP. Of course he thinks they are wonderful. The reality is they were both poorly thought out, and people have been shoring them up ever since. RTP used endless ports for no good reason. Nothing was symmetric. Bandwidth was considered of no consequence. It had no security, which has only recently been grafted on with sRTP. People have put massive effort into trying to live with or fix these things ever since. SIP started from the notion that call control is fundamentally simple, and H.323 was overly complex. SIP has been getting more complex ever since, and is now as complex as H.323. It had no security, and used an unreliable medium (UDP) for communication that needs to be reliable. Now networks are changing this, through a massive overhaul. To his credit Henning does accept that UDP for SIP was a dumb idea. He does say some good things, like: I consider the term soft switch a marketing term that, like its cousin, session border controller, seems to have no crisp definition that everyone can agree on. Even the Soft Switch Consortium has changed its name. The sooner we can drop the term soft switch from our discussions, the sooner people and their PowerPoint presentations will actually have to explain what their architecture is, rather than hiding behind an ill-defined label. Of Skype Most importantly, Skype got the out-of-the-box experience right-most of the time, it just works, without complicated configuration, even with NATs. and One of the nicer things about Skype is that they avoid being trapped in replicating the PSTN user appearance. Others have tried to make their software applications look like a cell phone or desk phone, which most often simply causes the software to inherit all the usability limitations that ISDN and other feature-rich phones had. A user interface stressing the buddy list and IM functionality seems a much better fit. and Also, they provided higher-quality audio codecs rather than feeling constrained by the notion that this would be wasted since the PSTN only supports narrowband audio. The technology to do this is fortunately readily available, both commercially and as open-source codecs like Speex, and SIP-based soft clients such as the new Yahoo Messenger and EyeBeam. He doesn't seem to really understand the strengths and weaknesses of IAX. IAX has drawbacks, but none of the problems he lists actually exist. Regards, Steve I won't debate Henning's comments on IAX2 other than to say it appears his comments reflect a less-than-full understanding of the protocol. I typically agree with Henning on many points when I read his RFCs and comments, but I suspect that on the topic of IAX2 he is going to be defensive as it is a protocol that has a distinctly pragmatic approach versus SIP which tries (too hard, IMHO) to be abstracted to allow any possible use. It is difficult to understand all of the possible variations of SIP's protocol flexibility, which makes programming somewhat difficult and makes testing regiments VERY difficult when attempting to build core systems which have to digest all possible flavors of SIP. The very nature of this complexity is why there is the hated Session Border Controller market in the first place. IAX2 is more easily understood due to it's more limited scope and regimented parsing, but it was not developed by committee. It has less flexibility, but... do I need anything more? Not typically. Voice, video, IM, HTML... sounds like a pretty full suite. The jury is still out on the benefits of IAX2 vs. SIP as a protocol which can be more widely used. Most SIP people of course will dismiss IAX2 completely, as it challenges the group-think. There are very few IAX2 developers, which is another struggling point for implementation of the protocol into commercial hardware. There is no BSD-licensed version of an IAX2 stack, to my knowledge, which again cripples IAX integration into products. It may be that SIP simply wins due to marketing, with IAX2's benefits and drawbacks never really being examined by most developers of applications. (Note to GPL-minded folks proposing a protocol: never, ever make the reference code GPL or LGPL. Make it public domain or BSD-licensed. GPL impedes commercial implementation for embedded code. Note I said impedes not prevents. But what adverb would you rather see describe your protocol uptake? impedes, prevents, confuses, or accelerates? You can make your application whatever license you want, but the _protocol_ stack implementation should be completely without restriction of uses.) I will strongly second Henning's comments on the
[Asterisk-Users] Best Voip provider
what is the best voip provider that provides good service ,good voice quality and good rates . any one have an experience with voip providers advice me. Regards; jonny Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] forward incoming analog call to SIP?
I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my lunch. Here are my various config files - maybe someone will take pitty on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel = 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] Transmit Silence=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid=Jane Smith 5678 host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=test include=local [sip] include=test include=local [test] 611,1,echo_test [local] exten = 1237,1,Dial(SIP/xlite1,10,t) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] forward incoming analog call to SIP?
Try changing [analog] to this [analog] include = test include = local exten = s,1,Answer ; Answer the call so we know its getting into * exten = s,2,Playback(transfer) ; Tell caller pbx is working exten = s,3,Dial(SIP/1237) ; transfer call to extension 1237 You have not allowed the ZAp software to dump the call into a dialplan it can deal with. In your setup the call comes in and is passed to the [analog] context but cannot be dealt with because the ZAP software doesn't know how to give the call to a particular extension. Adding the above line should cure that. Mark Dave Williams wrote: I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my lunch. Here are my various config files - maybe someone will take pitty on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel = 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] Transmit Silence=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid=Jane Smith 5678 host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=test include=local [sip] include=test include=local [test] 611,1,echo_test [local] exten = 1237,1,Dial(SIP/xlite1,10,t) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXO PCI Master abort
Dear Zaptel and wcfxo devellopers,
Hi, so far I have had no success moving this issue forward. Carl
Andersson has been kind enough to help build various kernels to try,
but with no success.
So, I have tried to debug the problem directly. So far I have applied
the patch below to wcfxo.c. (on the latest CVS head) This makes my
system stable again (in that I dont have to keep pressing the big red
button).
In principle I can not see how this patch would harm a working system,
and seems to me to be the right thing to do anyway. I would love to
know why not.
However, of course, this does not fix the problem.
Firstly the watchdog never re-starts the DMA engine (even with the
watchdog enabled in zaptel). This means that the driver sits there
impotent and in order to recover you have to unload and reload the
module.
Hardly ideal.
Second, of course, I am no nearer finding out why I am getting the
erroneous IRQ in the first place!
Any help much appreciated...
Cheers
Mark.
--- wcfxo.c 2005-08-13 14:16:44.690969912 +0100
+++ wcfxo.c.old 2005-08-13 08:51:00.0 +0100
@@ -726,18 +726,11 @@
/* Reset Master and TDM */
outb(0x01, wc-ioaddr + WC_CNTL);
outb(0x01, wc-ioaddr + WC_OPER);
-
-printk(UnMasking IRQ\n);
- outb(0x3f, wc-ioaddr + WC_MASK0);
}
static void wcfxo_stop_dma(struct wcfxo *wc)
{
- /* Enable interrupts (we care about all of them, except the one
that gave us the abort) */
-printk(Masking IRQ's, waiting for watchdog to restart\n);
- outb(0x2f, wc-ioaddr + WC_MASK0);
-
outb(0x00, wc-ioaddr + WC_OPER);
}
On 2 Aug 2005, at 13:56, Mark Burton wrote:
[posted here as well as -users as the situation is stranger than I had
first thought... and I'm running standard parts... ]
Hi, I have the following configuration, which doesn't seem to work,
any help much appreciated
I am trying to get a X101P FXO card working AT ALL!
(It has the Ambient chip on it)
All I get is:
FXO PCI Master abort errors.
Depending on the way it feels, either these are repeated till
/var/log/ is full, or I get one and then the thing hangs.
This may, or may not, have something to do with a message
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
I have tried all combinations of
Debian builds of Linux 2.4.27, 2.6.8 and 2.6.11
Debian builds of Zaptel
CVS builds of Zaptel
I have tried it on an old P2 and a newer P4
I have switch off asterisk to make sure it's not in the asterisk
configuration...
In all cases with the same result.
I've mucked with the IRQ's till they dont conflict.. no change...
I've tried 2 different cards
So, I'm clearly deluded as everybody else seems to have no problem.
Can anybody help - what silly thing have I done?
Cheers
Mark.
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[Asterisk-Users] [Asterisk-Dev] Re: FXO PCI Master abort
Dear Zaptel and wcfxo devellopers,
Hi, so far I have had no success moving this issue forward. Carl
Andersson has been kind enough to help build various kernels to try,
but with no success.
So, I have tried to debug the problem directly. So far I have applied
the patch below to wcfxo.c. (on the latest CVS head) This makes my
system stable again (in that I dont have to keep pressing the big red
button).
In principle I can not see how this patch would harm a working system,
and seems to me to be the right thing to do anyway. I would love to
know why not.
However, of course, this does not fix the problem.
Firstly the watchdog never re-starts the DMA engine (even with the
watchdog enabled in zaptel). This means that the driver sits there
impotent and in order to recover you have to unload and reload the
module.
Hardly ideal.
Second, of course, I am no nearer finding out why I am getting the
erroneous IRQ in the first place!
Any help much appreciated...
Cheers
Mark.
--- wcfxo.c 2005-08-13 14:16:44.690969912 +0100
+++ wcfxo.c.old 2005-08-13 08:51:00.0 +0100
@@ -726,18 +726,11 @@
/* Reset Master and TDM */
outb(0x01, wc-ioaddr + WC_CNTL);
outb(0x01, wc-ioaddr + WC_OPER);
-
-printk(UnMasking IRQ\n);
- outb(0x3f, wc-ioaddr + WC_MASK0);
}
static void wcfxo_stop_dma(struct wcfxo *wc)
{
- /* Enable interrupts (we care about all of them, except the one
that gave us the abort) */
-printk(Masking IRQ's, waiting for watchdog to restart\n);
- outb(0x2f, wc-ioaddr + WC_MASK0);
-
outb(0x00, wc-ioaddr + WC_OPER);
}
On 2 Aug 2005, at 13:56, Mark Burton wrote:
[posted here as well as -users as the situation is stranger than I had
first thought... and I'm running standard parts... ]
Hi, I have the following configuration, which doesn't seem to work,
any help much appreciated
I am trying to get a X101P FXO card working AT ALL!
(It has the Ambient chip on it)
All I get is:
FXO PCI Master abort errors.
Depending on the way it feels, either these are repeated till
/var/log/ is full, or I get one and then the thing hangs.
This may, or may not, have something to do with a message
Uhhuh. NMI received. Dazed and confused, but trying to continue
You probably have a hardware problem with your RAM chips
I have tried all combinations of
Debian builds of Linux 2.4.27, 2.6.8 and 2.6.11
Debian builds of Zaptel
CVS builds of Zaptel
I have tried it on an old P2 and a newer P4
I have switch off asterisk to make sure it's not in the asterisk
configuration...
In all cases with the same result.
I've mucked with the IRQ's till they dont conflict.. no change...
I've tried 2 different cards
So, I'm clearly deluded as everybody else seems to have no problem.
Can anybody help - what silly thing have I done?
Cheers
Mark.
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Re: [Asterisk-Users] forward incoming analog call to SIP?
Dave Williams wrote: I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my lunch. Here are my various config files - maybe someone will take pitty on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel = 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] Transmit Silence=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid=Jane Smith 5678 host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=test include=local [sip] include=test include=local [test] 611,1,echo_test [local] exten = 1237,1,Dial(SIP/xlite1,10,t) try this: [analog] exten = s,1,Answer exten = s,2,Dial(SIP/xlite1,10,t) exten = s,3,Hangup of course, this is an unconditional forward... -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] False Zap answer problem
And btw, I have CentOS 4.1. Could this be related with 2.6 kernel? Hi All, I am experiencing a very strange problem. I call the FXO channels (Zap/2 and 3) almost at the same time, and then hang both up. The operator extension is Zap/6, and after the greeting message Zap/6 starts to ring (there is no disconnect supervision here, and I disabled the busy detect for hangup detection, so the outside lines stay open until Asterisk hangs them up). As you can see on the following CLI section, when the VoiceMail promt starts to play for Zap/2, the system thinks that Zap/6-2 has answered the Zap/3, when in fact nobody answers it: -- Hungup 'Zap/6-1' -- Executing VoiceMail(Zap/2-1, u0) in new stack -- Playing 'vm-theperson' (language 'tr') -- Zap/6-2 answered Zap/3-1 I can very easily replicate this (did it 3 times). Where should I look for the source of this problem? Is it the TDM card or the Asterisk? How does Asterisk know if a line has been answered on a TDM card? This seems like some kind of cross-talk between the two FXO channels, but it's really strange. Has anybody had a similar problem? Of course, the workaround is to use some sort of hangup detection on outside lines, but I was having false hangup problems with busy detect, thus disabled it and noticed this problem (and see my above comment on disconnect supervision here). I have 2x TDM cards (version E/F) with 3x FXO and 4x FXS modules on each. Asterisk version is: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-10 22:55:45 UTC I have echocancel and echocancelwhenbridged enabled. (Could it be the echocanceller?) I would appreciate any help, Soner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why NAT problem
hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing not registered. i think asterisk is properly sending request to UA. any commentsthis sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved useragent SJLabs-SJphone/1.40.258 for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Announcement to called party
JP Carballo wrote What does the CLI say? Does it show Playing 'value-of-MES' (language 'en')? I'm using 1.0.8 here and I have no problems using A(x) in my dial strings in either ZAP or SIP channels. Yes, it does say Playing .. (language en) but there is no sound sent to the called extension Steven Hall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
On Fri, 12 Aug 2005, Matt Florell wrote: Short answer: NO Long answer: you have to send it to Digium for them to do an upgrade, they don't have an official process for this yet and won't give you a price, I have called and asked them many times. They also mention upgrades from your 405/410 to a 406/411 are available too, but again no specifics. Supposedly if you have a card with the 2nd gen firmware on it you can upgrade to the third gen firmware, whenever it would come out, in the field. Hmm.. That's funny. I called yesterday and talked to someone who told me the upgrade to Second Gen firmware was free, but that if I wanted to add the Echo Cancelling module, it would be $850. Since I do not have any major echo issues that software echo cancelling can't fix, I declined the upgrade. They even offered me an advance replacement option as long as I provided a Credit Card. The RMA process was painless. I spoke with Joy Lister. I should have my new card early next week. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Bruce Leetch wrote: Am I banging my head against at Windows/VMware/Linux/Asterisk incompatibility? Or can this work and I'm just doing something stupid (always a possibility with me). It's not going to work. Vmware presents a complete Virtual PC, so unless EMC / Vmware release drivers to specifically connect the Virtual PC to the real PC hardware, you are in trouble. The do a pass-through w/ USB, Serial and Paralell, but those are a different story. With a great amount of effort, I can drum up a spare machine, but I REALLY don't want to do this and would much prefer the VMware setup. Any advice will be welcomed. I'm afraid that under any Virtual platform (CoLinux, Vmware, MS Virutal PC) you are SOL as far as real access to hardware on the host PCI bus w/out special drivers written specifically for that purpose. On the other hand, I'm sure that Vmware would be happy to help you out if you gave them a couple of million bucks! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
Jonathan: Our provider continue selleing us SPA-841, if you want the contact, mail me outside the list. On 8/13/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Tom Rymes wrote: Chris, Maybe you could write a generic config file and post it to the wiki? I tried to post as a comment but the XML was excluded. How do I do that? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] yet another Asterisk and VMware question
On Fri, 12 Aug 2005, Tom Rymes wrote: VMWare is a virtual machine and has nothing to do with the physical layout of the box (which is why you can migrate vmware images across machines for example). If you want to run Asterisk under Linux setup a box to run it. Agreed. You would be better to grab a used $200 machine and install linux Asterisk on it. Unless you are scaling up to at least 10+ simultaneous calls, I would imagine that something you have lying around would handle it. If you insist on VMWare, I would imagine that you could configure a Sipura SPA-3000 to provide incoming (FXO) and analog extension (FXS) ports This works. I've done it on occasion for testing. However, because virtual PCs rarely operate on a real-time clock, mostly emulating these features, you will find that anything that read/writes to disk will suck badly. For example, it is nearly impossible to use the Voicemail features of Asterisk under Vmware, CoLinux or UserMode Linux. Believe me, I've tried! ;) This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs STABLE of 08/10 gcc4 issue
Hi all, I'm trying to make my cvs STABLE 08/10 srpms build properly on an updated FC4 box. When I rebuild the srpm with FC4's gcc4 I get this error: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O3 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-08/13/05-23:45:43\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -c -o channel.o channel.c channel.c:64: error: static declaration of 'uniquelock' follows non-static declaration include/asterisk/channel.h:58: error: previous declaration of 'uniquelock' was here make: *** [channel.o] Error 1 error: Bad exit status from /var/tmp/rpm-tmp.94064 (%install) When I force the use of gcc32 all is well and the asterisk srpm compiles fine. Any ideas how I can make asterisk compile with gcc4 too? Thanks and regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Push to talk and asterisk
I have a repeater using app_rpt, it seems to work just fine. Quoting Mustafa N. Deeb [EMAIL PROTECTED]: Has anyone been able to compile app_rpt? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Saturday, August 13, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Push to talk and asterisk You may be duplicating work that has already been done. http://www.zapatatelephony.org/app_rpt_article.pdf Mark, KC2ENI Mustafa N. Deeb wrote: Hi We are putting some efforts on having asterisk work as a PTT server over GPRS. Anyone interested to part of it , Please email me privately Best Regards Mustafa N. Deeb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11
Wich kind of E1 card do you use at the NORTEL ?? it was a PRI one??? witch model ??? On 8/12/05, Mark Phillips [EMAIL PROTECTED] wrote: Easily doable. I've done it twice now. Problem is that your users will never know they have messages waiting. Install a T1/E1 card into the * box and then use a T1 cross-over cable between the 2 boxes. Create a dialplan on the Meridian that points calls to the VM out over the new E1. As for forwarding the calls when busy or no answer, that's a little more tricky. You'll have to come up with some rules and numbers to allow the Meridian to decide what to do with those calls. In my case I wrote a forward on no answer and a forward on busy rule for every phone that needed VM. When you called ext 200 the call was sent to mailbox 2200 on the *. Users will have to get into the habit of calling the VM to check if there's messages. Hope that helps. Mark craz sead wrote: Hi all, Could somebody help me, i wanna connect asterisk for voice mail in the existing nortel pbx option 11 using e1 card ? anyone have a clue ? please help the conf. file thank all __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall will definately increase jitters in your voice conversation
Lokesh kumar wrote: Hi, If you will put firewall, then i think you will get high latency and consequently you will hear voice jitter in your conversation. so avoid putting firewall. Is this a troll or what? Anyway, there is a valid point here so I will address it as if it were not. The issue is that your perimiter control mechanisms can affect latency in any number of ways. Those of us who know what we are doing use them to *reduce* latency and *increase* sound quality by using various QoS traffic shaping technologies on the firewall. Similarly if you have a severely underpowered firewall and your firewall rules are overly complex then you very well could have sound quality degradation. However, it is a matter of firewall design and not really a matter of the class of technologies as a whole. Without a firewall w/QoS set up properly, I should be able to cause voice jitters by downloading, say, 10 Linux distro ISO's from various different locations... :-) The QoS system prevents this from causing problems :-) Best Wishes, Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall will definately increase jitters inyourvoice conversation
Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. But you still want a firewall here especially if you have several VOIP systems which could be making independent connections to the internet. The firewall in this case will hopefully not only do things like VPN for securing your data in trasit between your office and a remote one, but it will also provide a platform for QoS/traffic shaping. To avoid the firewall here is actually *asking* for sound quality problems in addition to the fact that you no longer have the entrence point to your network secured. Now to your point Almost any Linux system can be configured (if you know what you are doing) to perform all these firewalling functions. Just add an extra network card, put it on the perimeter of your network, set up iptables, traffic shaping, uninstall unnecessary software, use Netstat to doublecheck listening ports, etc. and you have your firewall. A firewall doesn't have to be expensive but some form of perimiter control is very helpful in these cases. Best Wishes, Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Voip provider (Broadvoice and Vonage comparison)
If you need a FXS, Vonage starts at $15. If you want to simply go soft-only, Broadvoice would be a better choice. After the marketing and all the features that nobody uses are thrown out, it comes down to consistency. Broadvoice has had some problems in the past 6 months, Vonage hasn't (that I know of). Vonage makes you throw away your ATA and won't let you reactivate it if you close your account. Broadvoice has free calling to US + 34 countries for a $25 account. I can setup a broadvoice account and have it up and running as quickly as I can click through the order process, login to get my password, and then setup the SIP config on my asterisk box. Vonage...not so much as you have to either activate a brand new retail unit or allow them to ship you one. Granted, you can add soft accounts once you have a hard account, but soft phone accounts are stuck at 500 free minutes. Choose your poison... No matter who you go with, there will be pros and cons. I guess by default, I choose Broadvoice, only because Vonage makes it hard to purchase only what I need. There are others out there too, cheaper, better, blah... Its not worth arguing over. They all have outages eventually, piss of a few active mailing-list'ers and suddenly their reputation goes to downhill. Vonage - if you are reading this, stop requiring people to buy a device just to get a softphone activated. Also, stop making us buy a NEW device if we decide to disconnect the old device for a while. While your at it, put an unlimited minutes option on your softphones. Look at Broadvoice's plans! Broadvoice - if you are reading this, please let me register with all your proxies, not just one! Also, stop answering my call just to play some corny message that says my call won't go through. I can reroute the call if you just congest the call invitation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny hashem Sent: Saturday, August 13, 2005 2:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Best Voip provider what is the best voip provider that provides good service ,good voice quality and good rates . any one have an experience with voip providers advice me. Regards; jonny Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vmail.cgi
I'm trying to get the vmail.cgi script to work. Followed the instructions in the wiki, but I'm getting stuck with this error: Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 96. I chmodded the files and directories used by vmail.cgi per the wiki instructions, but it appears Apache can't access anything oustide /var/www I'm running CentOS4/Apache ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation
On that note... IPSec tunnels seem to reek havoc on the echo canceling/training process. Anytime our Cisco PIX loads up, the echo complaints start coming in. Stay away from the IPSec tunnels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. But you still want a firewall here especially if you have several VOIP systems which could be making independent connections to the internet. The firewall in this case will hopefully not only do things like VPN for securing your data in trasit between your office and a remote one, but it will also provide a platform for QoS/traffic shaping. To avoid the firewall here is actually *asking* for sound quality problems in addition to the fact that you no longer have the entrence point to your network secured. Now to your point Almost any Linux system can be configured (if you know what you are doing) to perform all these firewalling functions. Just add an extra network card, put it on the perimeter of your network, set up iptables, traffic shaping, uninstall unnecessary software, use Netstat to doublecheck listening ports, etc. and you have your firewall. A firewall doesn't have to be expensive but some form of perimiter control is very helpful in these cases. Best Wishes, Chris Travers Metatron Technology Consulting ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation
Wiley Siler wrote: The question was not can I secure a Linux box without a hardware firewall. The question (or statement really) was will a firewall add jitter and lower performance. A good firewall architecture w/QoS will actually prevent jitter and increase performance, I might add. That answer is obviously a big NO. Can you secure a Linux (or even Windows) machine by closing ports? Sure. It helps immensely. However, an advantage of hardware is that you are physically separating the traffic from the end point. The analogy I would use here is that you could purchase a safe for each person in your house and have them each keep all their valuables in it, but it is often cheaper and easier to focus on securing entrence-points. The same is doubly true for office buildings, and also quite true for computer networks. I typically use used P1's running Linux for firewalls. They work great and have all the capabilities I need including QoS and secure management. Sure, all the ports closed on a Linux box can protect that machine. However, having only web (for example) traffic going to your Apache server is really beneficial. The server can focus on delivering pages and not spend any CPU cycles on is this a good packet? Should I drop it?. A firewall (software or hardware) should also be able to better deal with DOS and things of that nature. Port securing does nothing to assist with DOS. DOS doesn't include a TCP/IP stack does it? ;-) By Things of that nature are you including CP/M? Actually port securing can provide some measure of protection against DoS attacks in that fewer services are available to attack. However, you are correct that this protection is probably insignificant. So... You are totally right, you can secure a box that way. However, a firewall (be it software or hardware) is far superior a method. When you say software or hardware I assume you mean hardware like PIX and software like BlackIce. I am not sure where a stripped down Linux version running on a P1 which does firewalling and only firewalling fits in. I call that type of system a hardware firewall simply because it is a dedicated piece of hardware which does perimiter control and only perimiter control. Where VOIP is concerned, use a dedicated firewall system with QoS capabilities. Period. (Yes it is possible to run such a system on Windows, but I certainly don't advise it.) I prefer the hardware method myself as it is a matter of management and additional features. However, for some, a software method may be better. I ran Mandrake SNF (a shorewall implementation) for a long time so I have been there. Considering you can run a Linux firewall on a 386 machine worth $20 makes the fact that so many people don't have firewalls seem just ridiculous. Bear in mind that finding replacement parts (NIC's etc) for your 386 may not be trivial. That is why I use P1's with PCI slots... Also it is often impossible to get OpenGK to compile on such a machine due to memory limitations (my P1 firewall even has this problem and it has a whopping 32MB RAM). So the older you go, the less functionality you may be able to add. Best Wishes, Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] vmail.cgi
You might try to su - apache and make sure apache can read the file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Vega Sent: Saturday, August 13, 2005 5:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] vmail.cgi I'm trying to get the vmail.cgi script to work. Followed the instructions in the wiki, but I'm getting stuck with this error: Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 96. I chmodded the files and directories used by vmail.cgi per the wiki instructions, but it appears Apache can't access anything oustide /var/www I'm running CentOS4/Apache ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Quality
What is the optimum audio format and quality, codec, etc for using to play voice prompts in Asterisk? BTW - I am a Windows user, and about to record some prompts. Thanks Sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] (no subject)
TC ADSL should not bother PSTN as long as you use a proper filter. In our TC case a proper filter was supplied by the phone company when we installed TC the ADSL line. We Happily use Asterisk with an FXO card and an ADSL TC connection from the same phone line. Got a leviton DSL filter mounted in the wiring cabinet so I should be okay there. Glad I got it now, even though everything seems to work without it. Thx again for the info. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk forwarding confirmation?
Hi; I've been using Asterisk for a few months now, and I have run into an interesting issue that I thought someone else in the community may have run into: I have an Asterisk install set up to receive helpdesk calls, route them to several IAX extensions and an extension which is simply a forwarded call over the POTS to a cellphone, so that if no one is logged into their IAX extensions for whatever reason, the call would go to a cellphone. Ideally, I'd like it to move onto the next part of the dialplan if the cellphone isn't answered. Unfortunately, it turns out that most (if not all cellphones) have voicemail, which appears to Asterisk as though it had connected with a person, and it then connects the call. I had wanted to put in a small AGI application (or something similar) which asked for a single keypress to confirm that someone had actually picked up the phone call, but it seems as though using an AGI script would simply prompt the caller. Has anyone else had this sort of problem, and is there a way around other than creating call files and attempting to connect them with the incoming call? Thanks, Jeff Buchbinder ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Flash Transfer (callthrough)
Hello Everyone:)!,
I have a Incoming/Outgoing SIP Trunk setup to Broadvoice, is there a way to
send a Flash over the trunk, for example, to do flash transfers and
call-waiting?
I tried to use Flash() but it seems to not work on the sip trunk, my
configuration is as follows:
exten = 500,1,Goto(callthrough,s,1)
[callthrough]
exten = s,1,SetVar(NR=)
exten = s,2,Background(privacy-prompt)
exten = s,3,ResponseTimeout(10)
exten = s,4,WaitExten(20)
exten = _X,1,SetVar(NR=${NR})
exten = _X,2,Goto(s,3)
exten = *,1,Goto(s,1)
exten = #,1,Playback(transfer)
exten = #,2,Flash()
exten = #,3,SendDTMF(${NR})
exten = #,4,Flash()
exten = #,5,Hangup()
Debug output is as follows:
Please let me know, thanks! :)
Chris
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RE: [Asterisk-Users] yet another Asterisk and VMware question
This works. I've done it on occasion for testing. However, because virtual PCs rarely operate on a real-time clock, mostly emulating these features, you will find that anything that read/writes to disk will suck badly. For example, it is nearly impossible to use the Voicemail features of Asterisk under Vmware, CoLinux or UserMode Linux. Believe me, I've tried! ;) This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. Just since I am curious, what version of VMWare did you use and what kind of box where you running on? I've just moved my * box to a VM on ESX server and didn't play with voicemail until you mentioned it - now Allison's voice cuts in and out. Sounds like I am going to have to go back to the box I was running on previously. My original box is a P3 500 desktop while my VMWare ESX box is a dual P3 1.4GHz HP Proliant server. Rick The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why NAT problem
At firewall/NAT you have to do port forwarding. If your phone is at port 5060, NAT device will receive a connection and has to know that it is destined for your SIP phone. So, forward port 5060 to the phone. Rudolf - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 6:52 AM Subject: [Asterisk-Users] Why NAT problem hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing not registered. i think asterisk is properly sending request to UA. any commentsthis sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved useragent SJLabs-SJphone/1.40.258 for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo problem -- network related?
Rudolf Ladyzhenskii wrote: Hi, all I am running asterisk and my friends are running FireFly IAX phone. All is fine except one of them. When anyone tries to talk to him, tehre is a real bad echo. It is nothing to do with sound setup. Is he using a headset or speakers and microphone? Does he have Stereo Mix selected as a recording source? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues and Agent Call Logs/Wrapup logs
Now that we have a well functioning Asterisk system that queues our calls and distributes them to our CSRs, I would like to implement a better system for our agents to keep a log of all of their calls, which we currently do using MS Word. (As you would expect, this is a less than ideal solution!) I am looking for a simple program that will allow our agents to enter notes on each call that they take and save it to a database. Basically, The call would come in, and I would use something like astGUIClient or IPSwitchboard to perform a screenpop to a web page or program that would let the agent type in notes re: their call. Something along the lines of: Sally Jones called re: Account #1234 and wanted to know what her balance was and if she could make a payment by credit card. I took her information, charged the card and faxed her the receipt. Ideally, the note would be saved with the CDR data so I could search by extension, date, etc and find all relevant entries. Earlier in the month something similar was mentioned, and the poster mentioned searching Google, but I have yet to find an appropriate solution, and before I go and try to reinvent this wheel, I thought I would ask the list members if they have implemented something similar. Please let me know if you have any suggestions. Thank you for your consideration, Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
