Re: [Asterisk-Users] chan-sccp-easter2005 make error with stable 1.0.6?
Just installed and so far it works fine for a 30VIP, only issue is the Speed dials they dial out OK, but the cisco phone doesn't break dial tone even when the other party has answered. Do you have any suggestions Regards Robb ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Excternip and FWD
Hi I have updated from CVS about a week ago and got the externip working with FWD for outbound calls., but I'm having problems with inbound calls, I don't think they are even reaching the Asterisk box even though I have forwaorded 5060 and the rtp range specified, another thing I have notice, is that very ocasionally when I m using sip debug the internal IP is used for communication with FWD, Has anyone got inbound calls working with FWD and Externip through a Nat? Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX hard phone
Has any one seen or heard of the lastest developments fo the Farfon IAX phone? the web site Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
Brian West wrote: 1 dont start in debug mode.. you will get lost in all the garbage start with safe_asterisk then asterisk -r set verbose 4 call in to each number and watch for the DEBUG: lines after starting simple switch on blah Then it should print like DEBUG: 0 DEBUG: 0 DEBUG: 0 It might even print them 6 times instead of 3.. Ie if it prints DEBUG: 327 DEBUG: 0 DEBUG: 0 then that ring is 327,0,0 ( but to be safe lets round it to 5's) and make it 325,0,0 only have to hit within 10 +/- for it to match. bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks again Brian, one more question if i may ( soory for the hand holding) I've added the line below should the information show up when I am in asterisk gc, what do I have to do to get the correct info thanks again for all your help Robb Brian West wrote: Thats one thing that needs to be added to the patch.. I did this areound line 4471 in chan_zap.c ast_verbose( VERBOSE_PREFIX_3 DEBUG: %d\n, curRingData[counter1]); bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John Hi Had this patch working until I updated from CVS last week, now the x100p doesnt recognise the distictive ring again has the syntax changed for making this work? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!!! Having problems Starting Asterisk
I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!!! Having problems Starting Asterisk
I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk
Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try shutting down all * processes (including mpg123). Now, see if your audio works normally. If not, rmmod the zaptel/fx? modules, and see if that works. If not, you should start by getting your audio on the consloe to work normally first, then, check with the zap/etc modules loaded, then try * . One step at a time. Tim Thanks for the advice but I don't have any console audio device, I'm still working on it so any other advise would be appreciated, do you think I need to rebuild the system? Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 30VIP Phones
Hi Has anyone go the 30VIP phone to work with asterisk? If so how good us the usability of the Cisco 30VIP phone with asterisk either using chan_sccp or Chan_skinny? Thanks for your Help Robb -- Robert Boardman Tel:01617737929 FWD:86263 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk
John Fraizer wrote: Robert Boardman wrote: Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try commenting out all of the entries in musiconhold.conf. Then, stop asterisk. Then, do a pstree from the shell prompt and see if you still have some mpg123 processes hanging around. If you do, do a killall -9 mpg123 from the shell prompt. Then, start Asterisk. Look at the output of pstree again and make sure that mpg123 isn't being started. Something that I noticed was that there is a new set of mpg123 processes started for every class of musiconhold that you have specified. Making sure that mpg123 won't start at all will at least isolate if that is what is preventing Asterisk from starting properly. John -- who has NEVER been at the console while working with Asterisk but does have everything I want, including meetme musiconhold working after much work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks That worked Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incomming Distinctive ringing
Hi I have had distinctive ringing working before the patch was applied to the CVS tree, now it doesn't work, Could anyone point me in the right direction to debug distinctive ringing? Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_skinny
Hi All I have been working in getting 2 Cisco 12 sp phones working, they work fine over the lan, but I cannot get it to work across the Internet I only have one way voice Do es anyone have any advice Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 12SP
Hi Cullen You need to change the hard coded firmware in chan_skinny.c line 71 ish to the numbers that show up when your phone boots, re compile and all will be fine Robb Cullen Simpson [EMAIL PROTECTED] said: I am trying to get a Cisco 12SP phone to work with *. I do not have call manager. When start * and turn skinny debugging on I get this on the console: -- -- Starting Skinny session from 192.168.1.202 Recieved AlarmMessage Device SEP0010EB003E03 is attempting to register -- Device 'ipme' successfuly registered Requesting capabilities Version Request Received CapabilitiesRes Feb 23 16:29:29 WARNING[794722]: chan_skinny.c:2275 get_input: Skinny Client sent less data than expected. Feb 23 16:29:29 NOTICE[794722]: chan_skinny.c:2333 skinny_session: Skinny Session returned: Success --- The phone indicates that it is programming. The IP address of the phone is correct in the logs. Here is a snippet from my skinny.conf file: --- [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 192.168.1.11 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 ; allow = all ; disallow = ; Typical config for 12SP+ [ipme] device=SEP0010EB003E03 version=P002G204; Thanks critch context=outbound-analog line = 120 ; Dial(Skinny/[EMAIL PROTECTED]) --- Any ideas? -- Cullen Simpson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DPNSS and Asterisk
Hi Just one question do any of the Digium T1/E1 cards do DPNSS signaling? Robb -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI and Voicemail Memory increasing
Steven Critchfield wrote: On Wed, 2004-03-03 at 06:33, Robert Boardman wrote: Hi, With the current CVS as of last night 20:00GMT I was testing a asterisk with the e100p card using a PRI analyzer to excerise the 30 channels over and over, just going directly to voice-mail. Basically, I don't know what is going on but every time a PRI channel is picked up about 8K memory is used, but it is never released, I ran the test for about four hours last night, and th server started with 100Mb usage, and after 4 hours the memory usage had climbed to 270Mb. After some research and a few changes this is my current observation 1st Observation: After the memory leak was discovered I turned astmm on and recompiled *. The 'show memory allocations' and 'show memory summary' commands did not indicate that any of the * internals were growing during the test run. 2nd Observation: When I first boot the system (mod probes for zaptel and wct1xxp and * are run up from rc.local and there are no ISDN calls running) the memory is already leaking by 8Kb every few seconds. If I stop * and then restart it the memory leak stops until I start making ISDN calls... Does anyone have any advice to release the memory or stop the leak? Where are you getting the values for the memory usage? Are you just using the free memory on the system? yes, Im just really looking at TOP memory usage when asterisk is being hammered by the pri analyzer Is this just liunx using memory for buffers and cache?? Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco VIP30
Hi Just got a brand new Box Cisco VIP30 off ebay, the standard phone functions work fine, just a couple of questions, 1) how do I program the other buttons not on the standard keypad part.. 2) When I hang up the display doesn't clear and keeps the numbers just dialed on screen, can this be cleared down. thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail
Hi I have an asterisk voicemail system connected directly to a pri, there are no extensions connected to the asterisk box, anyway my questions is, can I get asterisk to call an associated phone number when a voicemail box has a message? Thanks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_sccp How-to
I wonder if anyone could post a how-to for the chan_sccp, I've downloaded and compiled the code, but I don't know where to go from here, any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK - 1471
In the UK we have a service that if you dial 1471, the last 6 calls are read out to you and you can pick which one you want by pressing 3, this means that 1471 shows in the cdr, has anyone created a script or an application that will read out the last callers and then dial the number? ( that they would like to share? I only ask before I start to re invent the wheel Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgettone 100 phone Configuration
Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All extensions busy
Hi Firstly could I thnk everyone who has helped me so far, I just have a couple of queries I have not had chance to debug this much yet but When using the tdm40p all extesions busy themselves out, and * cannot rint the extensions for incoming calls is this because I don't have a hangup statement at the end of the incoming context? if not has anyone any idea? does anyone have a quick and dirty IAX confiuration sample Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installing the wcfxs driver
Hi All when I modprobe the wcfxs drive and do a cat /proc/pci, it is sharing irq with my AGP and USB, I think this is causing the card to stop working, it would work for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto another IRQ in case it is this causing the problem Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the wcfxs driver
Hi My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, the IRQ to be used with a particular module? Robb Quoting Emanuele Pucciarelli [EMAIL PROTECTED]: On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote: for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto another IRQ in case it is this causing the problem You usually can, you should check your motherboard's documentation. I have an Asus MB and I can effectively disable IRQ sharing for the board in the setup area reachable at boot. Bye, -- Emanueel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] limiting out going calls to a maximum duration
I want to limit my sons phone useage, by setting a 30min limit on out going calls from his room is there a simple way of doing this with asterisk? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Grandstream power supplies..
When I was looking for a psu the only site I found for 5V ( and a decent price) was CPC, but I cannot remember the www address robb Quoting WipeOut . [EMAIL PROTECTED]: Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about a month!) and set it up.. This morning the power supply is dead.. I have looked for a new one online (In the UK using Maplin let me know if you know a better place.) becasue it would probbaly take too long to get one sent from China or the US and I need to get that station operational again.. There seem to be many choices for power supplies.. Looking on the bottom of the broken one it is a 5VDC 400mA output.. When I looked online for a new one the choices are for regulated, unregulated and switch mode power supplies with the regulated and switch mode ones being VERY much more expensive than the unregulated ones.. Which kind would do the job?? Also most of the variable output adapters are 4.5v or 6v, Not the required 5v.. Any help would be appreciated.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call Intrude
Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN TA
I have an ISDN TA that has 2 POTS interfases (FXS), can these be used with asterisk? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Caller ID and X100p
Hi I really need caller id to work in the UK, I understand that the X100p uses a US chipset,two questions 1) is that a product that converts UK to US caller id in line or 2) would it be possible to have modem that supports CID in parallel with the line and the x100p.The modem reads the line and reports the cid to asterisk Just thinking outloud Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dect Phone
Hi I have a problem with a new DECT phone I have bought The key pad works like a Mobile phone where you dial first then pick up the line, but it seems to dail too fast or spuriously, ie 012826736464 show on thew Asterisk console as 0012282677, could any one offer advice how to fix? Also when doing a ZAP bridge to this phone from an outside line the call is very echoy, but not an internal call, but only on this dect phone and no others any advice would be helpful Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ringing
Hi I've just signedup for Distinctive ringing on my PSTN line in the UK, could anyone explain what I need to add in the conf files to detect and route based on in comming Distinctive ringing Thanks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ringing
Does asterisk know when each ring comes in or just the first ring, ie so the cadence can be worked out? say over two rings? Robb Martin Pycko wrote: The X100P together with asterisk does not support the distinctive ringing detection on the line. Asterisk however can generate the distinctive ring over FXS ports. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recall doesn't seem to work
Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recall doesn't seem to work
Thanks for the advice Matteo but it didn't work, anthink else I may of
missed?
Robb
Brancaleoni Matteo wrote:
I forgot...
the main problem is that eu phones seems to have flash timings
~80 - ~120 ms , so with default zaptel values, a flash hook
('R' button) is received by asterisk as one pulse, since
the pulse time is set up to 150ms ...
Matteo.
Il sab, 2003-09-20 alle 01:15, Brancaleoni Matteo ha scritto:
Hi.
zaptel.h , line 789
#define ZT_DEFAULT_RXFLASHTIME 1250
For italy I had to lower it to 200,
also be sure to lower the pulse timer
(unless you're using a pulse phone with asterisk)
line 792
#define ZT_MAXPULSETIME (150 * 8)
I moved it to (20 * 8)
be sure not to set it under ZT_MINPULSETIME, that's (15 * 8)
Matteo
Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto:
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
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[Asterisk-Users] MY Sql CDR
Could someone point me in the right direction for setting up the mysql cdr function Thanks robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recall doesn't seem to work
Thanks that worked
Robb
Brancaleoni Matteo wrote:
mmh... have you enabled
threewaycalling = yes
transfer = yes
in zapata.conf ?
matteo.
Il sab, 2003-09-20 alle 10:51, Robert Boardman ha scritto:
Thanks for the advice Matteo but it didn't work, anthink else I may of
missed?
Robb
Brancaleoni Matteo wrote:
I forgot...
the main problem is that eu phones seems to have flash timings
~80 - ~120 ms , so with default zaptel values, a flash hook
('R' button) is received by asterisk as one pulse, since
the pulse time is set up to 150ms ...
Matteo.
Il sab, 2003-09-20 alle 01:15, Brancaleoni Matteo ha scritto:
Hi.
zaptel.h , line 789
#define ZT_DEFAULT_RXFLASHTIME 1250
For italy I had to lower it to 200,
also be sure to lower the pulse timer
(unless you're using a pulse phone with asterisk)
line 792
#define ZT_MAXPULSETIME (150 * 8)
I moved it to (20 * 8)
be sure not to set it under ZT_MINPULSETIME, that's (15 * 8)
Matteo
Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto:
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
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[Asterisk-Users] CPU Optimisations For asterisk
How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? Thanks for your Help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CPU Optimisations For asterisk
Thanks for your reply The system is a small installation but I was thinking about optimizations and wondered if there would be any particular benifit anyway thanks for the reply, your comments are very useful robb Quoting Alastair Maw [EMAIL PROTECTED]: Robert Boardman wrote: How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? CHOST=i686-pc-linux-gnu CFLAGS=-mcpu=athlon-xp -O3 -pipe Well, it might run slightly faster, but you probably won't really notice the difference. You might well be better off with -O2 rather than -O3, as -O3 tends to agressively unroll branches to inlines which reduces the amount of code that fits on the chip's cache, resulting in slowness. It's swings and roundabouts, really. If you're using echo cancelling, it should be quicker if you enable the MMX stuff for that (see the Asterisk Makefile). Why do you need the extra speed? If you're desperately trying to optimise things like this to gain extra performance, you must have a pretty big system. Pretty big system should mean you have the cash to upgrade your CPU a bit, which will make much more difference. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller Id AGI Script
As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the callerid for internal phones and cdr? I think it can be done with AGI but don't know where to start could someone point me in the right direction Thanks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller Id AGI Script
Hi Dave I've not completed the script yet, But you may not like this but I've had to use a win98 box for the zoom 3025C (important its the C model), the zoom modem is the only internal one I've found that can do uk caller id (but its not supported on the linux driver), and is still available. ( I bought it from PC world for £25) and I'm going to hack this script http://www.anderbergfamily.net/ant/caller id/ to produce the file required There are linux drivers for the card (www.linuxant.com) but the UK caller ID is not implemented (yet) but if lots of us post a reply to the message board and maybe they'll implement, I've already sent Linuxant the requested info http://www.linuxant.com/pipermail/hcflinux/2003q3/001056.html Robb Quoting Dave Wilson [EMAIL PROTECTED]: Robert, As you've already got the modem recording the callerid etc, I shall presume you're perl and bash knowledge should be up to scratch. All you gotta do is, using AGI and perl, run a perl script which parses the info from the file and simply use AGI - setcallerid to pass the number into asterisk. Good intro to AGI here http://home.cogeco.ca/~camstuff/agi.html AGI/perl related stuff here http://asterisk.gnuinter.net/ If you need some further help getting your script working, mail me offlist. Now, I've got some questions for you :) What modem model are you using and would you like to share your script which captures the uk callerid, as this has been a major shortcoming for asterisk with UK users. HTH Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd ringing conditions
I have two questions about incomming ring and extension ringing 1) When an incoming call is detected by asterisk it takes 2 or three rings before the internal phone ring does anyone know how I can fix this? 2) All internal phone ring on an incoming pstn call but after the call is answer all the other phone ring for a couple of tinkles how can I stop this from happening? Thanks for your advice robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Hi Areski it comes back with a blank page? Robb Areski [EMAIL PROTECTED] said: Can you try: http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03 And tell me about the result ! -Areski On Thu, 2004-03-25 at 11:13, Robert Boardman wrote: Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi and stream_file
Hi,
I trying to get agi with perl to stream a gsm file , and wait for a
digit , the agi gets to the stream but doesn't play back, could some one
explain how this works
here is a snip it of code
open(DAT,/etc/asterisk/1571.log) || die(Cannot Open File);
while( $sth-fetch() ) {
print DAT in while loop\r\n;
$AGI-stream_file('demo-echotest','6');
print DAT got past stream\r\n;
my $testfor = $AGI-say_number($clid, '36');
print DAT got past say number\r\n;
if ($testfor == 3) {
$AGI-exec('Dial', 'Zap/1/$clid');
exit;
}
};
$AGI-hangup();
$sth-finish();
$dbh-disconnect();
close(DAT);
the Got past stream never gets printed to the log file
Any advice would be appreciated
Thanks
Robb
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[Asterisk-Users] Compiling Zaptel 0.9.0 drivers
Hi I'm trying to compile the Zaptel Drivers, but I seem to be getting an error zaptel.c:131: warning: data definition has no type or storage class zaptel.c:132: error: parse error before config_must_be_included_before_module zaptel.c:132: warning: type defaults to `int' in declaration of `config_must_be_included_before_module' From the web it seems to be a problem with module.h ,which is in a folder /usr/src/linux/include/linux is this the correct folder for this file for asterisk ? Thnaks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small linux distro to run * in old boxes
Hi Victor I'm currently working in a Linux Distro, it is being internal alpha testing by my self and a couple of me my colleagues, over the next couple of weeks I'm hoping to release a beta version to the asterisk community., I'll keep you posted via asterisk users, about its features as it developed. The first to note is Its currently a 28Mb ISO for installation with asterisk installed with zaptel, and lib pri this includes apache perl PHP, and Mysql I will be producing a web site I post the address when it is ready Regards Robb Victor Perez wrote: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] recommend a Linux based TFTP server
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? if you need any more info just ask Robb Tony Hoyle wrote: David J Carter wrote: Where would I find cdr-csv? Usually in /var/log/asterisk The line looks funny because of the line breaks. zapata.conf ukcallerid=yes callerid=asreceived signalling=fxs_ks channel = 1 : BT line channel = 2 : Telewest line I also have immediate=yes, but that shouldn't affect anything. Are you sure you've updated the modules correctly (done make/make install, done an rmmod on the old zaptel module and a modprobe on the new one)? There isn't much to go wrong beyond that... if you run asterisk with debugging you'll get a log if it finds a callerID but it's basically the same that goes into the cdr-csv file. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: bt communicator`
Hi Peter, just a couple of quick questions if the my ethereal trace only shows [EMAIL PROTECTED] do I need the .brz? is the MD5sum in the ethereal trace, as I have compared all combinations of MD5sum with the ethereal trace and cannot see it any where? Still cannot register, any advice would be greatly appreciated Regards Robb Whisker, Peter wrote: Hi Robert; First, you have to use the SIP2 channel code (chan_sip2.c) from http://bugs.digium.com/bug_view_page.php?bug_id=759 as this does the proxy-authenticate properly. Get the module, follow the build instructions, and add noload=chan_sip.so to stop the old code loading. It will autoload the new one. You need to know the username that the Yahoo Communicator uses. Ethereal or similar will trace SIP for you. The username I type into communicator has .brz appended by the Communicator for some reason. The password is the one you type into communicator but I had to MD5 it. Comment below password shows how. [general] ;port = 5060; Port to bind to port = 5052 ; change to 5052 as 5060 will not authorise on BTCommunicator ; Note if you want local SIP on 5060, you need to use siproxd or similar to redirect (unless anyone knows otherwise) pedantic=no disallow=all; Disallow all codecs allow=alaw ; Allow codecs in order of preference ; BT uses a-law allow=ulaw allow=gsm ;allow=ilbc defaultexpirey=1200 ; Change for BT as it objects to 3600 - note deliberate spelling error register = [EMAIL PROTECTED]:[EMAIL PROTECTED] ; Need to state externip as the internal address otherwise BT won't work - something to do with NAT ;externip = 195.13x.xx.xx externip = 192.168.10.250 localnet = 192.168.10.0/255.255.255.0 ;. ;. ;. [bt] type=friend nat=yes disallow=all allow=alaw canreinvite=no username=[username].brz authuser=[username].brz fromdomain=btinternet.com fromuser=[username].brz auth=[username]:[EMAIL PROTECTED] ; I didn't have the .brz here and it works? md5secret=6eb36df5f5d94381973b6090b30e0f59 host=btinternet.com ;outboundproxy=sip.btcommunicator.bt.net;not needed ;outboundproxyport=5060 ;not needed ;MD5 ;alambil:/etc/asterisk# echo -n [EMAIL PROTECTED]:btinternet.com:[password] | md5sum ;6eb36fd5f5d94381973b6090b30e0f59 - Once this worked, I didn't change it. There are probably unneeded lines above. Regards Peter -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 09 October 2004 21:40 To: Whisker, Peter Subject: bt communicator` Hi Peter I have been following your post but didn't see the other emails about getting it working until now!! Could you please send me the details for the chan_sip2 method Thanks Robb This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off-topic: Avaya 46xx, release 032207 ... help
Cesc Santa wrote: Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I get on the firefox browser a connection has been reset error message. - Avaya admin guide keeps mentioning all the commands you can enter via the keyboard on the phone ... but they don't work for me ... (the MUTE + numbers combination). Any ideas? the web browser problem is the most annoying one. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users don't know about the web interface, but hold then RESET # resets the phone, HOLD ADDR # allows you to set the ip address etc try those Regards Robb ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Having problems posting to the list
Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
Fantastic, works brilliantly, this should be in the CVS source, it would be good if the MSN could be set and show up in the cdr? Robb Brian West wrote: 1 dont start in debug mode.. you will get lost in all the garbage start with safe_asterisk then asterisk -r set verbose 4 call in to each number and watch for the DEBUG: lines after starting simple switch on blah Then it should print like DEBUG: 0 DEBUG: 0 DEBUG: 0 It might even print them 6 times instead of 3.. Ie if it prints DEBUG: 327 DEBUG: 0 DEBUG: 0 then that ring is 327,0,0 ( but to be safe lets round it to 5's) and make it 325,0,0 only have to hit within 10 +/- for it to match. bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks again Brian, one more question if i may ( soory for the hand holding) I've added the line below should the information show up when I am in asterisk gc, what do I have to do to get the correct info thanks again for all your help Robb Brian West wrote: Thats one thing that needs to be added to the patch.. I did this areound line 4471 in chan_zap.c ast_verbose( VERBOSE_PREFIX_3 DEBUG: %d\n, curRingData[counter1]); bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 Port FXO cards
will this port sort out UK caller id? --- Original Message --- From: Mark Spencer [EMAIL PROTECTED] Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 Port FXO cards We *are* making progress, and i have a running prototype, however the production board is having some trouble going off hook, which is fairly important on an FXO interface! Mark On Wed, 19 Nov 2003, Surajee Ratnayake wrote: anyway, better if Digium can do it quickly, we are suffering a lot with channel banks, we need to replace these channel banks with 4 port cards - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:09 PM Subject: Re: [Asterisk-Users] 4 Port FXO cards Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [Asterisk] GSM access
Hi All Maybe this would be a beter solution, but you may have to buy directly from them http://www.artech.com.tw/html/gx100e/gx100e.htm Robb --- Original Message --- From: David Luyens [EMAIL PROTECTED] Sent: Mon, 24 Nov 2003 14:14:10 +0100 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: [Asterisk] GSM access Yes, call control is via serial rs232 and voice is analog interface. a couple of links where the interfaces are described for the siemens module: http://www.cnetek.net/zlxz/Interface_E/TC3x_Interface_v0310.pdf http://www.conigma.com/downloads/siemens/TC35T/tc35t_hd_01_v0300a_268766 .pdf David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Max Tulyev Verzonden: maandag 24 november 2003 12:06 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Re: [Asterisk] GSM access ÷ ÓÏÏÂÝÅÎÉÉ ÏÔ 24 îÏÑÂÒØ 2003 10:21 David Luyens ÎÁÐÉÓÁÌ: Almost evey GSM manufactor has these kind of modules. Ericsson: GM25, DM20,.. Siemens: TC35 (http://www.siemens-mobile.com/cds/frontdoor/0,2241,hq_en_0_2220_rArNr Nr NrN,00.html) And can it extract from GSM channel GSM encoded voice, just to not making recoding? -- ó õ×ÁÖÅÎÉÅÍ, íÁËÓÉÍ ôÕÌØÅ× (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grand stream phone and double nat
Hi I'm trying to configur a grandstream BT101 to connect to asterisk, both behind different NATs, I realise that a double Nat is a problem, I have tried using fwd forwarding to iaxtel as a solution but cannt seem to get them to connect as I think there is a codec problem as IAXTEL doesn't seem to accept alawor ulaw is this correct? Has anyone been able to connect a sip phone across a double NAT ? I realise this has been discussed before and sorry for that Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Client for P800/P900
Peer Oliver schmidt wrote: Nicolas Bougues wrote: On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote: Hi Guys, is there a client which can be used on the SonyEricsson P800/P900...? IAX would be cool, but i take anything that can connect (via bluetooth) to an asterisk-server ;-). The phone is Symbian, and can also execute java-stuff... Would be nice. Could turn any symbian based phone into a cordless IAX phone (with limited range, though). Is the P800 able to connect to a bluetooth AP ? Or maybe you have a bluetooth suite on your PC that is able to sense the presence of the P800 and enable the serial-over-bluetooth link automagically ? The P900 offers to be a VoiceGateway via Bluetooth. So, it looks as if it should be able to work the other way round, only. BTW: Nicolas, are you thinking of finishing up your SyncML tool (http://nicolas.bougues.net/syncml/) I Have been looking into this, and there is a sip client for the series 60 here, http://www.indtelesoft.com/buzz2talk/ I have been trying to get bluetooth to enable the ip stack but I'm not having much luck, so I haven't proved it yet works, oh yes any its currently half duplex Regards Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor fdiles
Hi, I have a load of files recorded with MixMonitor that are out of sync ie one leg of the call is 2-3 seconds behind the other, is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong Is it possible to edit the file and re sync the a/b leg? Thanks for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound PRI ISDN 30 problems
Hi All I'm having problems with outboud ISDN calls, They setup OK , and ring the other end OK, but when the call is answered I get a disconnect cuase 17 with an error message in the console of [Apr 15 08:06:13] DEBUG[4361] chan_zap.c: Found empty available channel 0/31 [Apr 15 08:06:13] VERBOSE[4601] logger.c: -- Starting simple switch on 'Zap/62-1' [Apr 15 08:06:13] VERBOSE[4361] logger.c: -- Accepting overlap call from '12345678901' to '0797' on channel 0/31, span 2 [Apr 15 08:06:13] VERBOSE[4361] logger.c: -- Channel 0/31, span 2 got hangup, cause 17 [Apr 15 08:06:13] WARNING[4601] channel.c: Unexpected control subclass '5' Any assistance would be greatly appriciated Regards Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN Call Droping only for outgoing
I have been trying to sort this out for a while now but with no luck I have isdn - asterisk- pabx on a te205 incoming calls work fine outgoing calls seem to work fine but the call is dropped when answered I think it is to do with the line [May 8 17:51:55] WARNING[4762] channel.c: Unexpected control subclass '5' that is causing the problem but I don't know how to fix, I think it is BT but they say the line is OK any advise would be greatly appreciated Thanks Robb [May 8 17:51:55] DEBUG[4711] chan_zap.c: Found empty available channel 0/31 [May 8 17:51:55] VERBOSE[4711] logger.c: -- Accepting overlap call from '0161555' to '0797355' on channel 0/31, span 2 [May 8 17:51:55] VERBOSE[4762] logger.c: -- Starting simple switch on 'Zap/62-1' [May 8 17:51:55] VERBOSE[4711] logger.c: -- Channel 0/31, span 2 got hangup, cause 17 [May 8 17:51:55] WARNING[4762] channel.c: Unexpected control subclass '5' [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:1] GotoIf(Zap/62-1, 0?8:) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/62-1, ) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:3] NoOp(Zap/62-1, ) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:4] Set(Zap/62-1, CDR(accountcode)=0797355) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:5] NoOp(Zap/62-1, ) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing [EMAIL PROTECTED]:6] Dial(Zap/62-1, ZAP/g1/0797355||r) in new stack [May 8 17:51:58] VERBOSE[4762] logger.c: -- Making new call for cr 32775 [May 8 17:51:58] VERBOSE[4762] logger.c: -- Requested transfer capability: 0x00 - SPEECH [May 8 17:51:58] VERBOSE[4762] logger.c: Protocol Discriminator: Q.931 (8) len=44 [May 8 17:51:58] VERBOSE[4762] logger.c: Call Ref: len= 2 (reference 7/0x7) (Originator) [May 8 17:51:58] VERBOSE[4762] logger.c: Message type: SETUP (5) [May 8 17:51:58] VERBOSE[4762] logger.c: [04 03 80 90 a3] [May 8 17:51:58] VERBOSE[4762] logger.c: Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) [May 8 17:51:58] VERBOSE[4762] logger.c: Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) [May 8 17:51:58] VERBOSE[4762] logger.c: Ext: 1 User information layer 1: A-Law (35) [May 8 17:51:58] VERBOSE[4762] logger.c: [18 03 a9 83 81] [May 8 17:51:58] VERBOSE[4762] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 [May 8 17:51:58] VERBOSE[4762] logger.c: ChanSel: Reserved [May 8 17:51:58] VERBOSE[4762] logger.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 [May 8 17:51:58] VERBOSE[4762] logger.c:Ext: 1 Channel: 1 ] [May 8 17:51:58] VERBOSE[4762] logger.c: [6c 0d 21 81 30 31 36 31 36 35 35 35 35 30 30] [May 8 17:51:58] VERBOSE[4762] logger.c: Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [May 8 17:51:58] VERBOSE[4762] logger.c: Presentation: Presentation permitted, user number passed network screening (1) '0161555' ] [May 8 17:51:58] VERBOSE[4762] logger.c: [70 0b a1 37 39 37 33 32 35 34 30 37 33] [May 8 17:51:58] VERBOSE[4762] logger.c: Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7973254073' ] [May 8 17:51:58] VERBOSE[4762] logger.c: [a1] [May 8 17:51:58] VERBOSE[4762] logger.c: Sending Complete (len= 1) [May 8 17:51:58] VERBOSE[4762] logger.c: q931.c:2881 q931_setup: call 32775 on channel 1 enters state 1 (Call Initiated) [May 8 17:51:58] VERBOSE[4762] logger.c: -- Called g1/0797355 [May 8 17:51:58] VERBOSE[4710] logger.c: Protocol Discriminator: Q.931 (8) len=10 [May 8 17:51:58] VERBOSE[4710] logger.c: Call Ref: len= 2 (reference 7/0x7) (Terminator) [May 8 17:51:58] VERBOSE[4710] logger.c: Message type: CALL PROCEEDING (2) [May 8 17:51:58] VERBOSE[4710] logger.c: [18 03 a9 83 81] [May 8 17:51:58] VERBOSE[4710] logger.c: Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 [May 8 17:51:58] VERBOSE[4710] logger.c: ChanSel: Reserved [May 8 17:51:58] VERBOSE[4710] logger.c:Ext: 1 Coding: 0 Number Specified Channel Type: 3 [May 8 17:51:58] VERBOSE[4710] logger.c:Ext: 1 Channel: 1 ] [May 8 17:51:58] VERBOSE[4710] logger.c: -- Processing IE 24 (cs0, Channel Identification) [May 8 17:51:58] VERBOSE[4710] logger.c: q931.c:3428 q931_receive: call 32775 on channel 1 enters state 3 (Outgoing call Proceeding) [May 8 17:51:58] DEBUG[4710] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 [May 8 17:51:58] VERBOSE[4762] logger.c: -- Zap/1-1 is proceeding passing
[asterisk-users] SERVICE CODES
Hi I'm trying to get the status of an extension that has DND set using the service code, or trying to disable the service codes altogether so that I can do them in the dialplan if needed any advice wout be appriciated Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
Olivier wrote: 2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? Regards Replying to myself, for an unknown reason, MWI is weirdly working : - Phone icon inconsistently shows awaiting voicemails, - NOTIFY message from Asterisk are still replied with 481 Call Leg/Transaction Does Not Exist When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail Notifications : - you can see SUBSCRIBE message - you can see NOTIFY answer - you can't see any 481 Call Leg/Transaction Does Not Exist reply to this NOTIFY message From then on, further NOTIFY messages are replied with 481 Call Leg/Transaction Does Not Exist and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db To: sip:sip:[EMAIL PROTECTED]:5060 http://sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520 Contact: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/0 (0/0) NOTIFY message rejected by S450IP (rejected means 481 reply) NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport From: asterisk sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED];tag=as5e574490 To: sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 96 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Voice-Message: 3/0 (0/0) The only difference I see between both is that new NOTIFY don't include : Subscription-State: active Do you see something else ? Is it possible to easily add this Subscription-State field without patching Asterisk source (I'm unable to do that) ? Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just worked out a good way of getting transfer working Using features .conf [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer DTMF A-D are valid DTMF signals but are not usually shown on standard phones so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
Olivier wrote: 2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Kevin P. Fleming wrote: Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. do you have a suggestion, there is only two fields that can be filled in that to refer to the R key, Application-type: I think this is content type Application-signal: what it sends? Hello, Reading this thread, I think I should have opened in the first place, 2 different threads as a common title is misleading to this R Hook-Flash key topic. Now, Gigaset S450IP base configuration web offers 2 fields to set R key : Application-type: Application-signal: When those 2 fields are respectively valued to Application-type: dtmf-relay Application-signal: 16 ... anytime this R-key is pressed, the base station would send a SIP INFO message to Asterisk. This SIP info is ended with : ... User-Agent: S450 IP02123000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/dtmf-relay Content-Length: 22 Signal=16 Duration=86 This 16 signal is interpreted as : Receiving INFO! * DTMF-relay event received: FLASH In my testing, I changed values like this Application-type: foo Application-signal: 16 2 and got a (single) SIP INFO message like this: User-Agent: S450 IP02123000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/foo Content-Length: 22 Signal=16 2 As Kevin told previously, Hook Flash transfer is not supported by Asterisk SIP stack. At the same time, it is written here (http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP) that : * Enable the R-button in SIP mode /fixed 14/09/2007/ So, what does this exactly mean ? Which values are to be typed in Application type and Application signal to make this R key be of any use ? Is it possible to pass several DTMF signals in a single SIP INFO so that Asterisk would receive a *2 anytime the R-key is pressed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'll reply to the correct thread [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfercapability DIGITAL
Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfercapability DIGITAL
yes and it is still set to speech I've even tried to port the old patch here http://bugs.digium.com/view.php?id=6251 to the system with no luck robb Melcon Moraes wrote: Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert boardman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:17:13 +0100 Delivered: Tue, 17 Apr 2007 19:15:09 Subject:[asterisk-users] Transfercapability DIGITAL Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176848736.557345.22480.arrino.hst.terra.com.br,4235,Des15,Des15 --Original Message Ends-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfercapability DIGITAL
Hi Chris I'm using Zap hardware , the second leg is always speech, and the far end anwsers and sets up a data call but there is no data transfered back so the call is dropped Regards Robb Christoph Fürstaller wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi robb, Have you just seen the bearer capability in asterisk or is the call nat working? I've seen that a digital call shows up as speech. You are using Zap? Or are you using mISDN? Cause there you have to set an extra parameter in the dial statement. chris... robert boardman schrieb: yes and it is still set to speech I've even tried to port the old patch here http://bugs.digium.com/view.php?id=6251 to the system with no luck robb Melcon Moraes wrote: Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert boardman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:17:13 +0100 Delivered: Tue, 17 Apr 2007 19:15:09 Subject:[asterisk-users] Transfercapability DIGITAL Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176848736.557345.22480.arrino.hst.terra.com.br,4235,Des15,Des15 --Original Message Ends-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Dipl.-Ing. Kurt Krenn - IT-Beratung Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 kkrenn (557366) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGJhXnR0exH8dhr/YRAoFkAJ0UEmz8y+XqLYqDhBTTDl7VbdEkjACfabkX X5mowtdnhs9qiX26oPxJJbA= =aBxY -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk and avaya SIP trunking
Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) [Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found. A SIP Debug of the packet when this happens on asterisk CLI is --- SIP read from 10.10.8.2:5060 http://10.10.8.2:5060 --- ACK Tel:+32564 SIP/2.0 Via: SIP/2.0/UDP 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 From: avayanew sip:[EMAIL PROTECTED];tag=d60c0430c7b26cbd To: Tel:+32564;tag=as51355066 Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 152795667 ACK Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO Content-Length: 0 Note: 10.10.8.2 http://10.10.8.2 is avaya and 10.10.8.1 http://10.10.8.1 is asterisk As I understand, we are getting a Tel URI and a + like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the 'Tel:+', so that the asterisk gets the number and thus follows the dialplan programmed in extensions file. Please advise. Any help is appreciated. Thanks as always Regards Krishna ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users you need to make sure the sip dial command in the ipoffice is set to dial 9n; feature dial code n in system the set the dial delay timer to 4 seconds and the dial delay count to 1 this will allow 4 seconds in between each digit there is a setting on the ipo to change the TEL:+ setting to url setting cannot remember wher it is but it in the sip trunk settings robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk and avaya SIP trunking
Krishna Sumanth Chava wrote: HI Robb, I had the checked the IP Office and i see that in the SIP Line Settings an option [checkbox] that says (Use Tel URI), which is unchecked. But i still get the Tel:+ in the SIP Header (even when it is turned on or off). you need to make sure the sip dial command in the ipoffice is set to dial 9n; feature dial code n do you mean that i need to program this in the ARS of the avaya IP office? i have version 4.1(9) firmware on the Avaya IP small Office. Can you share me on what Firmware version of avaya IP small Office, you got the Asterisk and avaya talking to each other. Thanks Krishna On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) [Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)? [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' rejected because extension not found. A SIP Debug of the packet when this happens on asterisk CLI is --- SIP read from 10.10.8.2:5060 http://10.10.8.2:5060/ http://10.10.8.2:5060 http://10.10.8.2:5060/ --- ACK Tel:+32564 SIP/2.0 Via: SIP/2.0/UDP 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 From: avayanew sip:[EMAIL PROTECTED];tag=d60c0430c7b26cbd To: Tel:+32564;tag=as51355066 Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 152795667 ACK Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO Content-Length: 0 Note: 10.10.8.2 http://10.10.8.2/ http://10.10.8.2 http://10.10.8.2/ is avaya and 10.10.8.1 http://10.10.8.1/ http://10.10.8.1 http://10.10.8.1/ is asterisk As I understand, we are getting a Tel URI and a + like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the 'Tel:+', so that the asterisk gets the number and thus follows the dialplan programmed in extensions file. Please advise. Any help is appreciated. Thanks as always Regards Krishna ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users you need to make sure the sip dial command in the ipoffice is set to dial 9n; feature dial code n in system the set the dial delay timer to 4 seconds and the dial delay count to 1 this will allow 4 seconds in between each digit there is a setting on the ipo to change the TEL:+ setting to url setting cannot remember wher it is but it in the sip trunk settings robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sorry its something like dial 9n; feature dial code n@192.168.0.1 where the ip address is the asterisk box robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * + Legacy PBX works but strange problem
Sriram wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via
span2)- legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which
is in sync with asterisk on span2)This works perfectly fine until
about 200 calls or so...After that time when asterisk tries to dial to
the legacy pbx - the call drops with error All are busy congested at
this time .the same is indicated on asterisk -rvv , but the
spans are up and active at that time... can anyone throw some light on
this ?
ZAPTEL.CONF
|
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
bchan=32-46
dchan=47
bchan=48-62
ZAPATA.CONF
|
|
context=pri-pstn
switchtype=euroisdn
pridialplan=local
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=1
pickupgroup=1
immediate=yes
musiconhold=default
signalling = pri_cpe
channel = 1-15
channel = 17-31
context=pri-legacy
immediate=yes
group=2
overlapdial=yes
signalling = pri_net
channel = 32-46
channel = 48-62|
| EXTENSIONS.CONF
|
|
;
; Context PRI-Public
;
[pri-pstn]
;
include = default
;
exten = s,1,Answer |
|exten = s,2,Dial(Zap/g2/1888); Dial to legacy pbx and sends the 4 DID
digits needed for the legacy pbx
exten = s,3,Hangup
;
; Context PRI-legacy
;
[pri-legacy]
;
include = default
;
exten = s,1,Answer
exten = s,2,DigitTimeout,2
exten = s,3,ResponseTimeout,2
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion|
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you need to pass the clock form the telco to the legacy pbx
ie
|span=1,1,0,ccs,hdb3,crc4|
Regards
Robb
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[asterisk-users] ISDN Cause codes
Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Cause codes
Some are mis dialed but most work one day but not the next they are all dialed manually Robb Don Kelly wrote: What is the source of the numbers you are calling? Are they previously-verified numbers from your database? Are some of them fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more than 3% of calls that I manually call. Have you researched some of the failures (examining the numbers that were attempted to be called)? I don't really see a problem with what you're reporting. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Cause codes
Thanks for the reply Could you be a little more specific? Thanks Robb Martin Smith wrote: Hi Robert, I'd suggest tweaking the Dial() arguments so that you (1) allow early audio, (2) don't force it play ringing to the calling party, and (3) modify any other options to be as relaxed as possible. if you make those changes, you'll start hearing the operator message recordings and those are sometimes easier to reference against the cause codes. Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Cause codes
I have found that the messages are not played as the hangup cause clears down the channel and passed hangup to the other end should I have progress() before the dial command? Robb Martin Smith wrote: Hi Robert, I'd recommend the following options for Dial() so that you corroborate operator messages w/ cause codes: 1. remove R and r - we've found this can supress operator recordings on early audio 2. likewise, remove m to disable MOH Also, check the values of DIALSTATUS to compare to HANGUPCAUSE. Good luck, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Friday, November 21, 2008 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN Cause codes Thanks for the reply Could you be a little more specific? Thanks Robb Martin Smith wrote: Hi Robert, I'd suggest tweaking the Dial() arguments so that you (1) allow early audio, (2) don't force it play ringing to the calling party, and (3) modify any other options to be as relaxed as possible. if you make those changes, you'll start hearing the operator message recordings and those are sometimes easier to reference against the cause codes. Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones
Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones
zap channel on one card to zap channel on another Robb Alex Balashov wrote: You mean a zap-to-zap call hairpinned into the same adaptor, or another one? Robert Boardman wrote: Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones
thanks Found that but sometimes I need to detect dtmf ie when playing back a recording Robb Philipp Kempgen wrote: Robert Boardman schrieb: I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? One (ugly!) solution is to change the DTMF tone frequencies in Asterisk so it doesn't recognize them any more: http://astrecipes.net/index.php?n=248 Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancelation
Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HFC Single port Cards
Hi all Been messing about with the single port cards for a number of years, but never got good results, I was thinking of giving them another go over Christmas and was wondering if anyone would share there recent experience, as to which driver works best MISDN BRISTUFF etc with the latest version of asterisk that supports Zaptel , I'll probably have a TDM400 card in the same box Thanks in advance Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warnings during a compile
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of âreadâ, declared with attribute warn_unused_result is this anything to worry about? can i safely ignore it? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warnings during a compile
On 04/02/2009 00:24, Mark Michelson wrote: Robert Boardman wrote: Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of âreadâ, declared with attribute warn_unused_result is this anything to worry about? can i safely ignore it? Thanks Robb I may be wrong about this part, but that class of warning is something that started appearing with a recent version of gcc (4.3 I think). Kevin Fleming took the time to clear up these warnings shortly after the release of this version of gcc, so if you are using a current checkout of Asterisk, you shouldn't see those warnings. In fact, looking at manager.c in my 1.4 and 1.6.0 checkouts, all calls to read(2) have their return value checked. To answer your question more directly, it's something that has a low potential to create problems, but given how long Asterisk had gone without checking those return values and how recently that was fixed, it's probably something you can ignore. Of course updating to a more recent checkout of Asterisk will clear such warnings up. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for your reply, I think I may need to use the 1.4..21.2 version as I'm still using zaptel for the pri card Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN30 Channels Locking
Hi Had an issue today where all channels connected to the telco when dialed returned WARNING[15366] chan_zap.c: Call specified, but not found? in the logs, when I removed the isdn cable and reinserted everything was fine any ideas? software Versions asterisk-1.4.21.2 zaptel-1.4.12.1 libpri-1.4.9 Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Data Modem
Jon Morgan wrote:
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data modem in order to maintain the internal phone system. Is
there any way we can configure our asterisk to allow them to dial in using
their modem?
Regards,
Jon.
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Hi jon
What system is it?
you need to set the transfer capability
eg
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?8:)
exten = _X.,2,Noop
exten = _X.,3,ringing
exten = _X.,4,set(CDR(accountcode)=${EXTEN})
exten = _X.,5,Noop
exten = _X.,6,dial(ZAP/g2/${EXTEN},,r)
exten = _X.,7,hangup
exten = _X.,8,Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,9,dial(ZAP/g2/${EXTEN})
exten = _X.,n,hangup
Regards
Robb
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[asterisk-users] Mix Monitor call quality
Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels Asterisk version 1.2.10 also under certain conditions Asterisk just stops any advice would be appreciated Thanks Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT Communicator (SIP???) and Asterisk
Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with Asterisk, or would someone be willing to help get this SIP provider to work? If you want more information about the BT communicator go to http://www.bt.com/btcommunicator/index.jsp just a quick run down of features 1) Home home rings BT Communicator rings (I think) 2) up to 5 different sip Users at once ie 5 extra home phone lines 3) Can dial almost any standard phone line ( no Premium or 1 numbers) based on you standard BT Tariff Thanks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk
gARetH baBB wrote: On Fri, 20 Aug 2004, Robert Boardman wrote: BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with Asterisk, or would someone be willing to help get this SIP provider to work? The only issue with it working with Asterisk is the current lack of reasonable Outbound Proxy support - or BT telling you where a direct SIP regitration server is (I've looked for one and failed). Otherwise it's easy, I've used Communicator with a range of the usual soft phones (X-lite etc.). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Gareth Heartened by your that you have got x-lite working, I have been trying, but failing to now get x-lite working, don suppose you could send me a quick screen shot of you x-lite settings? thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p
should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p
thanks for the reply Dan Does anyone know if the history buffer CID patch still works with the latest cvs? Robb Dan Tucny wrote: The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO modules due to the fact that the x101p is not capable of detecting polarity reversal events. Dan On Fri, 2004-09-10 at 17:38, Robert Boardman wrote: should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
Just done this for a client using an E1 Pri card in the avaya box and a sangoma a102, using qsig , works fine, I wouls recommend this to any oneits been up and stable for two months now Regards Robb housi mueller wrote: The main goal is that any extension from the Avaya PBX can make long distance calls using the asterisk server as VoIP gateway (using a SIP Provider). It would be also great if from a remote IP Phone (in an other location), a user could use the Asterisk server to dial in and the * forward’s the call to an Avaya extension. The Avaya has an VCM card an IP Phones (5610) as extensions. First I thought to connect the * to the Avaya through the ethernet interface but then I was reading in forums that there are for Avaya third party IP phone licence needed and that the communication with oh323 is not stable. I thought also putting the Asterisk in front of the Avaya. Telco T1 - Asterisk - T1 - Avaya PBX This could be a solution for later one. Right know for testing it would be to expensive. That's why I thought about the Avaya analog Asterisk FXO interconnection. Any suggestions..? */Thomas Kenyon [EMAIL PROTECTED]/* wrote: housi mueller wrote: I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBX’s. What sort of interaction are you after? It may be a better idea to try to intercept the line card with asterisk, or if the IP406 has a VCM card then to talk to it through the ethernet interface. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line
hi vincent, In the UK you can have multiple pots lines with the same telephone number. but you would need more fxo lines for this. Regards Robb Vincent Li wrote: Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it achievble in Asterisk to let them make PSTN call simulataneously through one sinle POTS line? I don't know anything about traditional PBX system, it seems one shop can have one single phone number and mutiple extensions, then the extensionss can make/receive PSTN call simultaneously, is this the same senerio as the one single POTS line to FXO and multiple extensions on FXSs? Thanks for help. Vincent ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Control playback
Hi All I have been asked if it is possible for an external application to be aware of the position of the playbcak of a file with control playback ie a file is playing and the user hits the fast forward button , is there a manager event that show how far into the file it has been played? thanks in advance Robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Tzafrir Cohen wrote: On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote: Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to make and receive DIGITAL type ISDN calls through the Asterisk box... but something in the latest generation of updates has broken it and although the calls seem to work the old PBX just won't route traffic. Voice calls still work fine. I've proven it's something in Asterisk by connecting the old PBX directly to our ISDN PRI line and it still works fine. What version is good? What version is bad? I have an outstanding problem with this,I have found that if you set overlapdial to no on the internal leg ie connected to the pabx it works, but you will have to set the pabx to dial en-block ie send all digits at once robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Tzafrir Cohen wrote: On Thu, Jan 03, 2008 at 12:24:38AM +, robert boardman wrote: I have an outstanding problem with this,I have found that if you set overlapdial to no on the internal leg ie connected to the pabx it works, but you will have to set the pabx to dial en-block ie send all digits at once Could you please be more specific? What versions have that problem? Could you provide some more details about your setup? heres the bug report http://bugs.digium.com/view.php?id=10941 Robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with zaptel and Udev
Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with zaptel and Udev
Tzafrir Cohen wrote: On Sun, Jan 13, 2008 at 05:33:58PM +, robert boardman wrote: Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Hmm is it udev that modprobes the modules on the PCI bus? yes I think it is , I'll re complie zaptel to see if that makes any difference Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with zaptel and Udev
thanks for the reply I'm already on 1.4.7.1 regards Robb Ed Nunez wrote: I had the same issue and updated my Zaptel drivers to version 1.4.17 and it's rebooting fine now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of robert boardman Sent: Sunday, January 13, 2008 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problems with zaptel and Udev Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA922
Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA922
Hi the asterisk version is 1.4.21.2 Here is the CLI -- Executing [...@incomming:1] Set(Zap/4-1, DB(lastcaller/zap4)=01942876818) in new stack -- Executing [...@incomming:2] GotoIf(Zap/4-1, 0?s-spoof|1:) in new stack -- Executing [...@incomming:3] Ringing(Zap/4-1, ) in new stack -- Executing [...@incomming:4] Set(Zap/4-1, CDR(accountcode)=s) in new stack -- Executing [...@incomming:5] Dial(Zap/4-1, SIP/105|20|tT) in new stack -- Called 105 Sip.conf ( with somethings changed) [gerneral] externhost=a.host.to.setup.com localnet=10.1.1.0/255.255.255.0 nat=yes [105] callerid=105 type=friend username=105 host=dynamic context=dialednum secret=red dtmfmode=rfc2833 disallow=all allow=alaw insecure=very ;mailbox=...@homr qualify=no nat=yes 2009/8/7 Danny Nicholas da...@debsinc.com Show us your CLI output. I suspect that you’re not getting a bridge and/or you’re timing out. Also sip.conf and user.conf would be helpful as well as Asterisk release. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *robert boardman *Sent:* Friday, August 07, 2009 9:01 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Linksys SPA922 Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium PRI cards for data usage?
Do you have to set aside kines for the data channels or can you have dynamic data channels, for example ISDN dialup internet backup? Robb 2009/9/1 Tim Nelson tnel...@rockbochs.com - Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 31 August 2009 21:59:28 Tim Nelson wrote: Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their Wanpipe software. Thanks for any pointers. You can. The keyword is nethdlc in /etc/dahdi/system.conf, although to enable it, you need to uncomment CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h and recompile the dahdi drivers. Once the active spans are configured with nethdlc, use the sethdlc command line utility to set up the bonded channels into the various network interfaces (hdlc0 through hdlcN). Depending upon your configuration, you may or may not also need to then configure the corresponding pvcN devices. Here is an article on the old Zaptel interface. While the name of the driver may have changed, the procedures remain the same: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ By the way, the method for determining which channels are bonded are as simple as the number of channels you configure together (on a single line) in /etc/dahdi/system.conf. For example, you can do as little as nethdlc=1 (for a single 64k channel) up to nethdlc=1-192 (for 8 T1s bonded into a single data device). Each nethdlc line in the config becomes a separate hdlcN device. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) Thank you Tilghman! That is exactly what I've been looking for! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] portech MV-378 SIP GSM Gateway
Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point during the call, the duration of the calls ate completely different call progression Call comes in from Zap channel and dials a mobile number on the prtech gateway and it dials out on sip trunk 103, the call progresses ok and after a time the call goes silent without any warning any advice would be greatly appriciated Regards Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Home line noise problem
I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
