Re: [asterisk-users] Cordless SIP Phones
Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn’t cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. It depends on the features you are looking for. The Aastra probably has the best support of VOIP features on the handset. The additional handsets are a little pricey ($99), and only four are supported (which is probably more than enough in most cases). If you don't care about VOIP features in the handset, the Uniden UIP1868 might be a good option. This can be purchased in an unlocked version from various online voip equipment sellers (e.g. www.voipsupply.com). The advantage of the Uniden set is that it uses the same handsets as their 5.8 Ghz cordless POTS phones, which means that you have a variety of handsets you can use (including a waterproof/submersible handset), and they are cheaper. The UIP1868 also supports up to 10 handsets (probably way more than you want, but the feature might be useful in some situations). There's also a much better chance you can buy a new or replacement handset a couple of years down the road. Otherwise, as others have suggested, you might consider just using an FXS adapter and using an analog cordless phone system. This is what I am currently doing (although I am seriously considering buying the UIP1868). The main problem I have with this solution is the delays introduced by caller id, especially if you want distinctive rings based on the caller. With an analog system asterisk has to send the caller id between the first and second ring. Then, at least with Uniden phones the handsets won't ring at all until the caller id has been received (if you've enabled the distinctive ringing feature). In my house the cordless phone typically doesn't start ringing until the SIP phones have already rung twice. One final note. It is possible to unlock the vtech 8100. Do a google search for cyt35. CYT is a program that was written to unlock various TI AR7 based devices, and it is known to work with the vtech 8100. I have no experience with this myself, but you might want to look into it if you still have the vtech 8100, and you are not planning on using it with Vonage. John ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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] SIP SDP keep original codec selection?
Hello all, When an incomming SIP call is reveived I would like to force Asterisk to keep the SDP codec selection for the resulting outgoing call to the destination SIP endpoint. Does anyone know how this could be acheived? I know that the allowed codecs for each SIP endpoint can be restricted in the sip.conf but need this to be dynamic based. Thanks Ian. ___ --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] parsing extensions
Hi all,
is where a possibility for simply parsing and changing variables for bad
characters ?
eg. removing a '/' from a number dialed by a manager-connected application
changing 123/4567890to 1234567890
via bash you could simply use 'echo ${exten/\//}' but i couldn't find a
working solution for the asterisk-extensions.conf
best regards
Dirk Rieger
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[asterisk-users] licence quick question
Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? best regards Thomas ___ --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] Re: NAT: RTP Path Optimization
PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is PC working fine in my Setup, but I want Extern1 to talk to Extern2 PC directly whitout going over Asterisk as the uplink is slow. PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? PC I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this PC doesn't help. PC Ideas, how to handle Extern-2-Extern (RTP bypass Asterisk)? Do I PC have to adjust nat somwhere? Set canreinvite to no for Int1/2. /Benny ___ --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 very slow when internet down
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, For that I've set up a local DNS Cache (on the asterisk) - maradns. And entered 127.0.0.1 as the first DNS Server in d/etc/resolv.conf. To decrease the time asterisk is trying to do a dns lookup, I've added this options to /etc/resolv.conf: options timeout:2 attempts:1 Chris... Paul Hales schrieb: Sadly, people have reported this fairly regularly. An option is to hard-code the server and IP address in your hosts file, but this can be even worse when the provider changes the IP address of the server... later, PaulH On Thu, 2007-01-25 at 22:27 +1030, Peter Mitchell wrote: Has anyone seen this issue with asterisk running like a dog when the internet is down ? Internal calls, incoming ISDN calls etc all seem to be affected. There is a local DNS server that is always available so I’m not sure why asterisk is so unresponsive. I’ve seen this on two different systems, and on 1 of them I commented out my SIP providers in sip.conf and it ran ok again. Thanks Peter. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.410 / Virus Database: 268.17.10/651 - Release Date: 24/01/2007 ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.1 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFvdhkR0exH8dhr/YRArthAJoCClafg82Xa+yJ4Mrk5c4Fx4os6QCeMRY2 8WHJK20Aac8ZdQrcM0OmHww= =lIFj -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
Re: [asterisk-users] licence quick question
Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter: Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? IANAL, but in my understanding - a dialplan is not code, but a configuration file - that is not affected by the GPL - You will have to hand out the dialplan as a file on the Asterisk server, else it is pretty useless - if your customer has shell access to the * machine, he could just read that file - AGI programs in script (perl,python,php,bash) will have to reside on the * machine as well, accessible by users with shell access - AGI binaries probably can be called separate programs in respect to GPL - they will be called by *, but are not imminently necessary nor binary-linked to Asterisk nor do they necessarily use asterisk libraries (but you should check which libs you link into your program, especially the licensing conditions for the asterisk-specific interface which might have a special license) In my understanding this means that as long as you do not change anything in the asterisk codebase but restrict yourself to configuration files and AGI programs, there is no need to disclose the code of those to your customers. They will very well have the right to obtain a copy of the Asterisk source code as such, or the Linux kernel source. You probably should tell them the machine runs GPL'ed code and hand them a copy of the GPL, and if requested, refer them to download sources of the source code used. BR Anselm ___ --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] Transfer on RTP timeout?
On 01/28/07 18:52 Florian Overkamp said the following: Nokia seems to have done something like this in their E-series (E60 etc) with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ? i think that's a FMC (fixed mobile convergence) client which both avaya and cisco wrote for the E series platform. my stock E61 doesn't have such a client, though it has the SIP 2.0 symbian client. as for the original poster, what you can probably do is to trap the hangup, and perhaps modify app_dial.c to set the hangup cause in DIALSTATUS for RTP timeouts, then take appropriate redialling action as part of the h extension. do note that this is off the cuff, and i'm not sure how difficult it'd be to do this. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] licence quick question
On Mon, Jan 29, 2007 at 12:30:47PM +0100, Anselm Martin Hoffmeister wrote: Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter: Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? IANAL, but in my understanding - a dialplan is not code, but a configuration file - that is not affected by the GPL Unless it is based on GPLed dialplan, of course. The same applies to all the others. E.g: I figure that the sample dialplan that is distributed with Asterisk is distributed under the same license as Asterisk itself. - You will have to hand out the dialplan as a file on the Asterisk server, else it is pretty useless - if your customer has shell access to the * machine, he could just read that file or use #exec and your own top secret obfuscated binary. For the client to figure out from 'show dialplan' (if they have root access to the system or access to the manager interface). - AGI programs in script (perl,python,php,bash) will have to reside on the * machine as well, accessible by users with shell access - AGI binaries probably can be called separate programs in respect to GPL - they will be called by *, but are not imminently necessary nor binary-linked to Asterisk nor do they necessarily use asterisk libraries (but you should check which libs you link into your program, especially the licensing conditions for the asterisk-specific interface which might have a special license) AGIs are not linked with Asterisk. This has been explicitly clarified. Same goes for the AMI. In my understanding this means that as long as you do not change anything in the asterisk codebase but restrict yourself to configuration files and AGI programs, there is no need to disclose the code of those to your customers. Right. They will very well have the right to obtain a copy of the Asterisk source code as such, or the Linux kernel source. You probably should tell them the machine runs GPL'ed code and hand them a copy of the GPL, and if requested, refer them to download sources of the source code used. Right. Forgetting such basics is a silly reason for many of the GPL license violations that shouldn't have happened in the first place. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Polycom Provistioning Issue
Jason, Email me off-list and I will ship you a pack of usable configs. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 26, 2007, at 3:48 PM, Jason Walker wrote: Fixed that issue but it does not change the error 0126204105|cfg |3|00|Image sip.ld has not changed 0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 0126204105|cfg |3|00|Downloaded application image is identical to current version 0126204105|cfg |3|00|Phone successfully provisioned 0126204136|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36 2007 William M. Conlon wrote: Looks like the network time server isn't provisioned. -- Bill 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). ___ --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 ___ --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] Re: Heartbeat on Digium T1 PCI cards?
Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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
[asterisk-users] Pickup() ringing extension and call waiting
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten = s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include = outbound-local
include = parkedcalls
exten = 200,1,Dial(SIP/somebody1|20|tTrR)
exten = 201,1,Dial(SIP/somebody2|20|tTrR)
exten = 202,1,Dial(SIP/somebody3|20|tTrR)
exten = _8.,1,Pickup(${EXTEN:1})
[outbound-local]
ignorepat = 9
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
When there is incoming call and extension 200 rings, I press 8200 to pickup a
call and I get disconnected.
here is debug from asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, SIP/somebody1|60|tTrR) in new stack
-- Called somebody1
-- SIP/somebody1-081bea58 is ringing
-- Executing Pickup(SIP/somebody3-081b3cd8, 200) in new stack
== Spawn extension (internal, 8200, 1) exited non-zero
on 'SIP/somebody3-081b3cd8'
Also,
Call waiting seems to not work. While having a conversation I hear beep in
my phone but the M2 (second line) button doesn't blink so I can not pickup
second call and put first one on hold.
from my zapata.conf:
[channels]
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callerid=asreceived
usecallerid=no
hidecallerid=no
threewaycalling=yes
transfer=yes
callwaiting=yes
cancallforward=yes
;;hanguponpolarityswitch
busydetect=yes
faxdetect=both
group=1
callgroup=1
pickupgroup=1
context=incoming
channel = 3
;;channel = 4 ;; no line yet
Any ideas?
Thank you in advance,
Dominik
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Re: [asterisk-users] Pickup() ringing extension and call waiting
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten = s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include = outbound-local
include = parkedcalls
exten = 200,1,Dial(SIP/somebody1|20|tTrR)
exten = 201,1,Dial(SIP/somebody2|20|tTrR)
exten = 202,1,Dial(SIP/somebody3|20|tTrR)
exten = _8.,1,Pickup(${EXTEN:1})
[outbound-local]
ignorepat = 9
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
When there is incoming call and extension 200 rings, I press 8200 to pickup a
call and I get disconnected.
here is debug from asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, SIP/somebody1|60|tTrR) in new stack
-- Called somebody1
-- SIP/somebody1-081bea58 is ringing
-- Executing Pickup(SIP/somebody3-081b3cd8, 200) in new stack
== Spawn extension (internal, 8200, 1) exited non-zero
on 'SIP/somebody3-081b3cd8'
Pickup works on a channel, not on an extension number, so in the above
example you effectively execute
Pickup(200)
but need to have mapped the 200 so that you do
Pickup(SIP/somebody1)
Regards,
Steve
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Re: [asterisk-users] Pickup() ringing extension and call waiting
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote: Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. Call waiting seems to not work. While having a conversation I hear beep in my phone but the M2 (second line) button doesn't blink so I can not pickup second call and put first one on hold. disclaimerI have never used D-Link SIP phones/disclaimer Where there is only one SIP registration on the phone, Asterisk cannot have any control over the presentation of lights on the phone - This is entirely up to the handset to manage. One potential workaround is to provide multiple SIP logins (one per line) and arrange for asterisk to call the second line if the first is busy (use incoming_limit SIP settings perhaps?). This may or may not have the desired effect on that phone model. Cheers, Steve ___ --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] Pickup() ringing extension and call waiting
On Monday 29 January 2007 03:20:16 pm Steve Davies wrote:
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I
would like to be able to pickup ringing extention from any SIP phone
using Pickup() application.
from my dial plan:
[incoming]
exten = s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include = outbound-local
include = parkedcalls
exten = 200,1,Dial(SIP/somebody1|20|tTrR)
exten = 201,1,Dial(SIP/somebody2|20|tTrR)
exten = 202,1,Dial(SIP/somebody3|20|tTrR)
exten = _8.,1,Pickup(${EXTEN:1})
[outbound-local]
ignorepat = 9
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
When there is incoming call and extension 200 rings, I press 8200 to
pickup a call and I get disconnected.
here is debug from asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, SIP/somebody1|60|tTrR) in new stack
-- Called somebody1
-- SIP/somebody1-081bea58 is ringing
-- Executing Pickup(SIP/somebody3-081b3cd8, 200) in new stack
== Spawn extension (internal, 8200, 1) exited non-zero
on 'SIP/somebody3-081b3cd8'
Pickup works on a channel, not on an extension number, so in the above
example you effectively execute
Pickup(200)
but need to have mapped the 200 so that you do
Pickup(SIP/somebody1)
Regards,
Steve
What do you mean by mapping the 200 ?
In this example I can pickup any ringing extension:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
If phone with number 42 rings you can catch the call by dialing 742. You
don't need to use the context
exten = _7.,1,Pickup(${EXTEN:1}) works for all contexts.
Regarding call waiting, internally when I'm having a conversation and someone
calls me, then my second line button blinks and I can pickup a second call
putting first one on hold. Problem just with real call waiting from PSTN.
Thanks,
Dominik
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[asterisk-users] Rxfax and txfax
somebody know how to compile the rxfax and txfax apps under asterisk 1.4.0?? i get this errors: Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_rxfax.c - app_rxfax.o app_rxfax.c:60: warning: data definition has no type or storage class app_rxfax.c:60: warning: type defaults to 'int' in declaration of 'STANDARD_LOCAL_USER' app_rxfax.c:62: warning: data definition has no type or storage class app_rxfax.c:62: warning: type defaults to 'int' in declaration of 'LOCAL_USER_DECL' app_rxfax.c: In function 'phase_e_handler': app_rxfax.c:105: error: 't30_stats_t' has no member named 'column_resolution' app_rxfax.c:105: error: 't30_stats_t' has no member named 'row_resolution' app_rxfax.c:116: error: 't30_stats_t' has no member named 'row_resolution' app_rxfax.c:122: error: 't30_stats_t' has no member named 'row_resolution' app_rxfax.c: In function 'phase_d_handler': app_rxfax.c:147: error: 't30_stats_t' has no member named 'columns' app_rxfax.c:147: error: 't30_stats_t' has no member named 'rows' app_rxfax.c:148: error: 't30_stats_t' has no member named 'column_resolution' app_rxfax.c:148: error: 't30_stats_t' has no member named 'row_resolution' app_rxfax.c: In function 'rxfax_exec': app_rxfax.c:247: warning: implicit declaration of function 'LOCAL_USER_ADD' app_rxfax.c:281: warning: passing argument 1 of 'fax_init' from incompatible pointer type app_rxfax.c:281: error: too many arguments to function 'fax_init' app_rxfax.c:284: warning: assignment discards qualifiers from pointer target type app_rxfax.c:287: warning: assignment discards qualifiers from pointer target type app_rxfax.c:304: warning: passing argument 1 of 'fax_rx' from incompatible pointer type app_rxfax.c:307: warning: passing argument 1 of 'fax_tx' from incompatible pointer type app_rxfax.c:344: warning: passing argument 1 of 'fax_release' from incompatible pointer type app_rxfax.c:350: warning: implicit declaration of function 'LOCAL_USER_REMOVE' app_rxfax.c: At top level: app_rxfax.c:356: warning: no previous prototype for 'unload_module' app_rxfax.c: In function 'unload_module': app_rxfax.c:357: error: 'STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_rxfax.c:357: error: (Each undeclared identifier is reported only once app_rxfax.c:357: error: for each function it appears in.) app_rxfax.c: At top level: app_rxfax.c:363: warning: no previous prototype for 'load_module' app_rxfax.c:368: warning: no previous prototype for 'description' app_rxfax.c:374: warning: no previous prototype for 'usecount' app_rxfax.c: In function 'usecount': app_rxfax.c:376: warning: implicit declaration of function 'STANDARD_USECOUNT' app_rxfax.c: At top level: app_rxfax.c:382: warning: no previous prototype for 'key' make[1]: *** [app_rxfax.o] Error 1 make: *** [apps] Error 2 regards rene ___ --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] Simple question
The first include references another context within extensions.conf. Contexts are defined by words in brackets. In your example, there would be a context in extensions.conf that would look like: [inbound] Contexts allow for setting up difference services and difference user capabilities all within the extensions.conf file. The second include is including the contents of multiple *.conf files located in a directory called inbound. JB [EMAIL PROTECTED] 1/27/2007 6:50 AM Whats the difference between the following statements in extensions.conf include=inbound AND #include inbound/*.conf -- Regards Rizwan Hisham Software Engineer - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: Rhino cards lock up system -- anyone else ever seen this?
Turns out this appears to be related to hald -- the hardware abstraction layer daemon running on Centos. I had the almost identical situation occur with a completely separate system which I loaded Trixbox up on, with a single Digium TDM400P card in it. Struggled for several hours over the weekend trying to figure it out. I ended up shutting off all the services I didn't specifically need, and turned them back on one at a time (turned out, hald was the first I tried -- I was most suspicious with it). As soon as I started it up, it locked the system up. Turned all the other services back on, leaving hald off, and the system is running fine. Did the same with the original problem system, and now have no problems with the Rhino cards either! On 1/23/07, Barry D. Hassler [EMAIL PROTECTED] wrote: Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a modprobe rxt1, but once you run ztcfg -vvv, the system will lock up within a few seconds, no errors reported in logs or console. I'm stumped, Rhino is stumped, and I haven't seen any other threads of this nature. -- Barry D. Hassler -- Barry D. Hassler ___ --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] Re: NAT: RTP Path Optimization
Benny Amorsen schrieb: PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is PC working fine in my Setup, but I want Extern1 to talk to Extern2 PC directly whitout going over Asterisk as the uplink is slow. PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Asterisk gives out his own public adress and stays in the Media-Path between internal and external Phones. This avoids NAT-Problems. PC I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this PC doesn't help. PC Ideas, how to handle Extern-2-Extern (RTP bypass Asterisk)? Do I PC have to adjust nat somwhere? Set canreinvite to no for Int1/2. But then all RTP Traffic of my internal phones will go over Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern and Extern-to-Extern should go P2P and Intern-2-Extern should go over Asterisk, see picture ___ --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] put Agi script in queue
Hi everyone dou you know if is possible to put an Agi script in a queue? For Example 1 - Caller joins the queue 2 - Agi script starts ... ... Agi script ends 3 - Hangup. Is it possible? thanks ___ --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] LookupCIDName / LookupBlacklist syntax
WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is
deprecated.
Please use ${DB(cidname/${CALLERID(num)})} instead.
[WARNING[8384]: app_lookupblacklist.c:104 lookupblacklist_exec: LookupBlacklist
is
deprecated. Please use ${BLACKLIST()} instead.
I seem to be unable to find any update to the new syntax for these functions in
1.4. Does
anyone have the syntax or can someone point me in the right direction for
this ?
Thanks
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Re: [asterisk-users] Does X100P decode caller ID?
Leo Ann Boon wrote: It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. Leo ___ --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 02:08.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Intel Corporation Digium X100P/X101P analogue PSTN FXO interface Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr+ DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 20 Region 0: I/O ports at 3000 [size=256] Region 1: Memory at f7ce (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- cat /etc/zaptel.conf # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us fxsks=1 # cat zapata.conf [channels] busydetect=yes busycount=7 relaxdtmf=no callprogress=no callreturn=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=1.0 txgain=-1.0 immediate=no context=house:in signalling=fxs_ks callerid=asreceived channel = 1 faxdetect=both faxdetect=incoming faxdetect=outgoing faxdetect=yes useincomingcalleridonzaptransfer=yes CallerID works fine with my X100P :-P signature.asc Description: OpenPGP digital 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
[asterisk-users] SIP + short numbers + name of customer
We are using a couple of Grandstream GXP2000 SIP-phones with Asterisk. In our dial-plan, we have implemented a list of short numbers in extensions.conf, like: exten = 1234,1,Dial(Zap/0987654321) So when I pickup the SIP-phone, and I dial 1234, the system dials 0987654321 and connects me to that customer. Unfortunately I cannot see the name of the customer, and I do not know if perhaps I punched the wrong short number. Is there a way to have Asterisk print the name of the customer on the SIP-phone display, instead of 1234? Maybe the implementation (see above) is not optimal, and there is better way to deal with these short numbers? The same question for incoming calls: it would be great to have Asterisk print the name of the customer on the display when a call comes in, instead of his phone number 0987654321. Z. ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
I failed to notice that it was included in 51363 - I just checked, and that change is indeed already in. Sorry, my mistake. I generally do not change the -march setting, so I am probably using an i386 default. Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] SIP + short numbers + name of customer
Zoilo Gomez wrote: So when I pickup the SIP-phone, and I dial 1234, the system dials 0987654321 and connects me to that customer. Unfortunately I cannot see the name of the customer, and I do not know if perhaps I punched the wrong short number. Is there a way to have Asterisk print the name of the customer on the SIP-phone display, instead of 1234? Maybe the implementation (see above) is not optimal, and there is better way to deal with these short numbers? In general the answer to his is No. However, many SIP phones support a directory on the phone, if you put in a directory entry into the phone for that customer then that info should show up when you dial. Remember the GS BT101 CANNOT display anything except numbers on its display. The same question for incoming calls: it would be great to have Asterisk print the name of the customer on the display when a call comes in, instead of his phone number 0987654321. This is the default if Asterisk gets Caller*ID with the call. On PRI the Caller*ID NAME is sent a moment after the call setup. A Wait(1) or Wait(.5) as the first priority in the dialplan for incoming calls may make Asterisk receive the Caller*ID information. ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
On Mon, 29 Jan 2007, Steve Davies wrote: I failed to notice that it was included in 51363 - I just checked, and that change is indeed already in. Sorry, my mistake. I generally do not change the -march setting, so I am probably using an i386 default. I get segfaults with the VIA C3 and C7 chips (on CN1000 and other EPIA boards) with I leave it as the defaults. I need the -i586 option. -i686 seems the be the default in the makefile. I understand it's to do with the MMX instructions used in some of the codecs... Gordon Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 ___ --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] Disconnected Calls
I upgraded to the newest 1.2 Zaptel release and this is still occurring. I checked and the digium card is not sharing an IRQ with any other devices. I also changed busycount=8, and set callprogress=no. The call drops are still occurring. Mid-conversation ` in 10 calls will be disconnected. Any other suggestions? This is a relatively low volume system. Usually running less than 1 or 2 concurrent calls. Would turning on debugging logs to a file cause a problem? Many thanks, Ejay Hire -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire Sent: Wednesday, January 24, 2007 3:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Disconnected Calls Hello. I am running asterisk 1.2.14 on a Dell poweredge with a Digium FXO/FXS card connected to 6 analog lines and using Linksys spa942 phones. My users are complaining of randomly disconnected calls, and when I watch the log (debug warning,notice,error), I don't see any cause. It looks like asterisk is seeing a hangup from the analog end. I have attached my zaptel.conf and zapata.conf. What additional information can I provide to make this an intelligent question? Many Thanks, Ejay Hire Zapata.conf ; Zapata telephony interface [trunkgroups] [channels] musiconhold=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=1200 busydetect=yes callgroup=1 pickupgroup=1 immediate=no group=0 language=en context=default rxgain=12.4 txgain=4 signalling=fxs_ls rxwink=300 ; Atlas seems to use long (250ms) winks relaxdtmf=yes channel = 1 channel = 2 channel = 3 channel = 4 channel = 5 channel = 7 channel = 8 group=1 channel = 6 Zaptel.conf cat /etc/zaptel.conf # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsls=1 fxsls=2 fxsls=3 fxsls=4 fxsls=5 fxsls=6 fxsls=7 fxsls=8 loadzone= us defaultzone = us ___ --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] International dialing with GPX-2000 and early dial
Other phones have a defined dialplan, just like an ATA the GXP is the only
phone I've seen like that!
I had a sudden stroke of genius, I haven't tested it, but I'm sure it would
work. Define a DISA with no password at extension 011, and define a context
where international calls can be dialed without 011, IE:
exten = 011,1,DISA
[gs-intl]
exten = _xx.,1,Dial(ZAP/g0/011${EXTEN})
and then asterisk can handle the timeouts
On 11/20/06, Anthony Kepler [EMAIL PROTECTED] wrote:
We are on the same page.
If you happen to find a solution - or know of a way that other phones
address these issues, please let me know.
Andrew Joakimsen wrote:
Ok, I actually GOT a GXP-2000. It does not have a dialplan. You
cannnot dial without the handset off-hook. I do not seem to find a way
to use early dial for international calls in a practical way, not
being able to dial international calls is not acceptable. Having to
dial # or send for domestic calls isnt either, and neither is having
to wait 4 or 5 seconds for domestic calls to complete
Or am I missing something?
On 11/8/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Early dial is a feature on the phone that makes use of the 484
(Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance
dialing.
I'm really hoping to find a best-of-both-worlds solution to this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own dialplan? Use that and disable
early dial
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am trying to allow users to place outgoing international
calls
from a
GPX-2000 with early dial enabled, connected to Asterisk
1.2.12.1 http://1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance,
Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new
stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single
digit to
represent the .
What I need is for it to wait a reasonable amount of time for
additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the
following
in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5)
in
new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the
dial out
attempt.
I assume there must be something relatively obvious I'm
missing
here...
if anyone can shed some light on this, it would be greatly
appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED] |
SIP/Email
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Re: [asterisk-users] ATCOM AT 468 manuals and firmware anyone?
Erick Perez wrote: both not available. but thanks. Email edwin ( [EMAIL PROTECTED] ), he will be able to help. There is a newer firmware available than the one on their website (v4.2b5) which fixes problems with freezing and introduces a phone book, a digitmap and simple dialplan. There are a few other ATAs out there that use the CS6220 (iirc leadtek make one) so you can probably get firmware images that'll work from other sources too. If the worst comes to the worst, I've got some firmware images for it here somewhere. (most recent looking once called sip_ag468_vr42.r0 ) As for the manual, I've you'll probably need to get it from edwin, setting the unit up with asterisk is trivial. On 1/28/07, Leif Neland [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --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
[asterisk-users] internal and external interfaces
Before adding a second interface to my asterisk box I'd like to get some feedback on having and internal interface with a private address and external interface with a public interface. You know like pros, cons, configuration suggestions, and anyone's true experience trying such a design. I have one concern. Is asterisk aware enough to respond to traffic with the correct ip address? In other words if a packet enters the asterisk box over the private interface and the necessary routes are in place for return traffic respond over the private interface will asterisk keep the private address in its replies for all types of traffic? You may respond to me offline if you'd like. I appreciate any feedback. Thanks, -CP ___ --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] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
If it's using RBS then 56k is the right number. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Saturday, January 27, 2007 12:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] max tnt pri voice channels 56k or 64k,does it matter, selection parameter? Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007 11:11 AM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.14/657 - Release Date: 1/29/2007 9:04 AM ___ --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] Installed TDM02B - Problem when other end hangs up
Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. When the problem occurs, calling that ZAP line from outside seems to reset it as well. I'm sure it's something obvious I've overlooked, but I'd appreciate any pointers. Thank you, -- Warm Regards, Lee ___ --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] LookupCIDName / LookupBlacklist syntax
something like (AEL syntax):
if (${DB_EXISTS(cidname/${CALLERID(num)})})
CALLERID(name)=${DB(cidname/${CALLERID(num)});
Derek Whitten wrote:
WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is
deprecated.
Please use ${DB(cidname/${CALLERID(num)})} instead.
[WARNING[8384]: app_lookupblacklist.c:104 lookupblacklist_exec: LookupBlacklist
is
deprecated. Please use ${BLACKLIST()} instead.
I seem to be unable to find any update to the new syntax for these functions in
1.4. Does
anyone have the syntax or can someone point me in the right direction for
this ?
Thanks
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Re: [asterisk-users] Installed TDM02B - Problem when other end hangs up
Hello, Do you include in your zapata.conf answeronpolarityswitch=yes hanguponpolarityswitch=yes There are any problems with hang up Regards On 1/29/07, Lee Jenkins [EMAIL PROTECTED] wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. When the problem occurs, calling that ZAP line from outside seems to reset it as well. I'm sure it's something obvious I've overlooked, but I'd appreciate any pointers. Thank you, -- Warm Regards, Lee ___ --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] Heartbeat on Digium T1 PCI cards?
If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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 ___ --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] International dialing with GPX-2000 and early dial
I have been down this path with Grandstream but they (for reasons I don't
understand) want to upgrade the firmware to have a dial plan.
So the best you can do is use early dial, for all fixed length numbers in
the * dial plan this works reasonably well. International numbers vary in
length so apart from trimming the digit time-out there not much you can
do.
The GXP 2000 is a great phone is it's a pity that they don't want to
develop e the phone to make it even better.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Other phones have a defined dialplan, just like an ATA the GXP is the only
phone I've seen like that!
I had a sudden stroke of genius, I haven't tested it, but I'm sure it
would
work. Define a DISA with no password at extension 011, and define a
context
where international calls can be dialed without 011, IE:
exten = 011,1,DISA
[gs-intl]
exten = _xx.,1,Dial(ZAP/g0/011${EXTEN})
and then asterisk can handle the timeouts
On 11/20/06, Anthony Kepler [EMAIL PROTECTED] wrote:
We are on the same page.
If you happen to find a solution - or know of a way that other phones
address these issues, please let me know.
Andrew Joakimsen wrote:
Ok, I actually GOT a GXP-2000. It does not have a dialplan. You
cannnot dial without the handset off-hook. I do not seem to find a way
to use early dial for international calls in a practical way, not
being able to dial international calls is not acceptable. Having to
dial # or send for domestic calls isnt either, and neither is having
to wait 4 or 5 seconds for domestic calls to complete
Or am I missing something?
On 11/8/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Early dial is a feature on the phone that makes use of the 484
(Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance
dialing.
I'm really hoping to find a best-of-both-worlds solution to
this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own dialplan? Use that and
disable
early dial
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am trying to allow users to place outgoing international
calls
from a
GPX-2000 with early dial enabled, connected to Asterisk
1.2.12.1 http://1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance,
Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new
stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single
digit to
represent the .
What I need is for it to wait a reasonable amount of time
for
additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the
following
in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5)
in
new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the
dial out
attempt.
I assume there must be something relatively obvious I'm
missing
here...
if anyone can shed some light on this, it would be greatly
appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED] |
SIP/Email
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Re: [asterisk-users] Heartbeat on Digium T1 PCI cards?
Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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 ___ --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] T1 Wire Level Tapping
Wow, thanks for the awesome reply :) On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. There are a number of systems using ISDN digital taps. The proper way requires a high impedance bridge - you don't want to load the line that you're tapping. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: You can directly bridge the 2 ports and extract what you need as you bridge - see pridump.c in libpri. You don't even need asterisk, just the zaptel and libpri. The only problem with this approach, is that the bridge becomes a point of failure. Your box down, your PRI goes down as well. S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. This is technically correct, but I don't know how well it works. Eicon recommends a similar technique to do monitoring with their Eicon Server cards. For the BRI, it's done this way. But for the PRI card, they actually suggest using a custom cable. Eicon cards have a special Hi-Z monitoring mode to support this application. http://www.eicon.com/worldwide/solutions/How_To_Call_Tapping_and_Monitoring_with_Diva_Server FYI, Voicetronix has a Hi-Z version of their OpenPRI card that work with an open-sourced voice logging application available from their site. Leo ___ --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] T1 Wire Level Tapping
I am very interested in the DACs capabilities of Digium cards, there is no information anywhere on this. I could always do pri bridging via libpri like you suggest however. But having hardware handle the bridging onboard a single PCI card would help reduce my server requirements for a final product, as long as I can spy on active channels somehow. I don't think its going to work that way, I wil test out libpri for a bit. Shane+ On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. There are a number of systems using ISDN digital taps. The proper way requires a high impedance bridge - you don't want to load the line that you're tapping. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: You can directly bridge the 2 ports and extract what you need as you bridge - see pridump.c in libpri. You don't even need asterisk, just the zaptel and libpri. The only problem with this approach, is that the bridge becomes a point of failure. Your box down, your PRI goes down as well. S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. This is technically correct, but I don't know how well it works. Eicon recommends a similar technique to do monitoring with their Eicon Server cards. For the BRI, it's done this way. But for the PRI card, they actually suggest using a custom cable. Eicon cards have a special Hi-Z monitoring mode to support this application. http://www.eicon.com/worldwide/solutions/How_To_Call_Tapping_and_Monitoring_with_Diva_Server FYI, Voicetronix has a Hi-Z version of their OpenPRI card that work with an open-sourced voice logging application available from their site. Leo ___ --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] Zap channels staying offhook - restart required
Try setting AbsoluteTimeout() as the first parameter in your dialplan entry. Check it out on voip-info.org On 1/28/07, kjcsb [EMAIL PROTECTED] wrote: Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --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] parsing extensions
Hello,
Check app_backticks - it is an external application which should be
compiled on your system.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
[EMAIL PROTECTED] wrote:
Hi all,
is where a possibility for simply parsing and changing variables for
bad characters ?
eg. removing a '/' from a number dialed by a manager-connected
application
changing 123/4567890to 1234567890
via bash you could simply use 'echo ${exten/\//}' but i couldn't find
a working solution for the asterisk-extensions.conf
best regards
Dirk Rieger
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Re: [asterisk-users] Installed TDM02B - Problem when other end hangs up
Carlos Rojas wrote: Hello, Do you include in your zapata.conf answeronpolarityswitch=yes hanguponpolarityswitch=yes There are any problems with hang up I tried adding these parameters as you suggested, but then was unable to dial out at all. Removing them allows me to dial out again, but still experiencing the same problem as before. Thank you, -- Warm Regards, Lee ___ --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] H.264 *Not Patented*
On 27 Jan 2007, at 16:33, Lee Jenkins wrote: Although I wouldn't complain about a free G.729 codec, I have to be honest in saying that $10.00 isn't that great of an expense considering the better call quality you get. Does G.729 work by pushing up the compression, therefore moving from work from the Network to the CPU ? Across my LAN, I'd probably be able to handle *fewer*, rather than more calls across my * exchange if this was the case. If it's cleverer that this, I think I'll have to speculate a few dollars, assuming my Snoms can talk in G.729. cheers -a -- Regards, Andy Davidson http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. After playing around a bit, it appears that this is just random as far as I can see. It may allow me to dial a few times, but then hangup. After rebooting my server, it may let me dial once and then start hanging up. I really hope it's not a fight with every TDM that I will have to install. After reading so many problems posted on the list, I thought I had got off easy, lol. -- Warm Regards, Lee ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Lee Jenkins wrote: After playing around a bit, it appears that this is just random as far as I can see. It may allow me to dial a few times, but then hangup. After rebooting my server, it may let me dial once and then start hanging up. I really hope it's not a fight with every TDM that I will have to install. After reading so many problems posted on the list, I thought I had got off easy, lol. I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. -- Warm Regards, Lee ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Lee Jenkins wrote: Lee Jenkins wrote: After playing around a bit, it appears that this is just random as far as I can see. It may allow me to dial a few times, but then hangup. After rebooting my server, it may let me dial once and then start hanging up. I really hope it's not a fight with every TDM that I will have to install. After reading so many problems posted on the list, I thought I had got off easy, lol. I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Do you have exten = _. ANYWHERE in your config. If so, that could cause an issue like this. ___ --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] Installed TDM02B - Problem when other end hangs up
Hi, Lee: Lee Jenkins wrote: Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. After playing around a bit, it appears that this is just random as far as I can see. It may allow me to dial a few times, but then hangup. After rebooting my server, it may let me dial once and then start hanging up. I really hope it's not a fight with every TDM that I will have to install. After reading so many problems posted on the list, I thought I had got off easy, lol. This problem is very common. I am in Alberta and a Telus customer. I have a very similar issue: When the remote party hangs up, the TDM card does not detect the disconnect. Sometimes it sits on the line for more than 30 seconds, making it impossible to make incoming or outgoing calls on the channel. This is the configuration in zapata.conf: ; define channels context=incoming signalling=fxs_ks channel = 4 I use kewlstart signalling, which is supposed to cover most every situation, right? I've called the telco to discuss this issue, but nobody has a clue -- be it about call disconnect signalling, analog to PBX connections or anything else for that matter. This morning I spent more than 30 minutes on the phone and got transferred through to four departments, all of whom assured me that the *next* department would be able to answer my question. One person I spoke with insisted that most everyone with a PBX is using digital lines, but I know firsthand that lots of people still use analog lines with their PBXs. They must be doing something to clear the lines after the other caller hangs up. Any ideas? -Stephen- PS: For what it's worth, Verizon owns 33 percent of Telus; the network equipment in BC is pretty much the same as what is in use in what used to be GTE's regions. ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. The problem is simply that call disconnect information is not being passed on somewhere. Either 1. The carrier is not signalling (possible -- I find that call disconnect signalling is spotty, at least in Alberta) 2. The signal is peculiar and the card doesn't recognize it. In my case I can't say definitively one way or the other, and the repair staff at Telus have been no help whatsoever. I should say that this pretty much makes the TDM cards useless -- in these parts, anyway -- unless you use forward on busy; and then it severely limits the flexibility because of the latency between when a call terminates and the channel becomes available again. -Stephen- ___ --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] Installed TDM02B - Problem when other end hangs up
Carlos Rojas wrote: Hello, Do you include in your zapata.conf answeronpolarityswitch=yes hanguponpolarityswitch=yes This doesn't work everywhere. I don't think Verizon does disconnect signalling with a polarity switch, though I'd be happy to be corrected. What part of the world are you in, Carlos? -Stephen- ___ --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] Installed TDM02B - Problem when other end hangs up
Eric ManxPower Wieling wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on the caller first.
Then it will dial out again. Odd.
Do you have exten = _. ANYWHERE in your config. If so, that could cause
an issue like this.
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought (and posted incorrectly above) that I could dial into the
system from outside and reset the line, but that does not always work so
it really seems random to me. I think the only thing that definitely
resets the line so that it will work is rebooting the server unfortunately.
--
Warm Regards,
Lee
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Re: [asterisk-users] Installed TDM02B - Problem when other end hangs up
Stephen Bosch wrote: Hi, Lee: Lee Jenkins wrote: Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. After playing around a bit, it appears that this is just random as far as I can see. It may allow me to dial a few times, but then hangup. After rebooting my server, it may let me dial once and then start hanging up. I really hope it's not a fight with every TDM that I will have to install. After reading so many problems posted on the list, I thought I had got off easy, lol. This problem is very common. I am in Alberta and a Telus customer. I have a very similar issue: When the remote party hangs up, the TDM card does not detect the disconnect. Sometimes it sits on the line for more than 30 seconds, making it impossible to make incoming or outgoing calls on the channel. This is the configuration in zapata.conf: ; define channels context=incoming signalling=fxs_ks channel = 4 I use kewlstart signalling, which is supposed to cover most every situation, right? I've called the telco to discuss this issue, but nobody has a clue -- be it about call disconnect signalling, analog to PBX connections or anything else for that matter. This morning I spent more than 30 minutes on the phone and got transferred through to four departments, all of whom assured me that the *next* department would be able to answer my question. One person I spoke with insisted that most everyone with a PBX is using digital lines, but I know firsthand that lots of people still use analog lines with their PBXs. They must be doing something to clear the lines after the other caller hangs up. Any ideas? -Stephen- Yes, that seems to be the same boat I am in and I am at a loss as to how this can be worked around as well. Regardless, its unacceptable and I am supposed to start putting systems in the ground here in a few weeks with TDM cards. Not looking forward to wrestling with these kinds of problems. Would be nice to just have something (anything) I.T. related just work without hassle for once ;) Hopefully someone better experienced than us will chime in and give us some guidance. -- Warm Regards, Lee ___ --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] Installed TDM02B - Problem when other end hangs up
Stephen Bosch wrote: Lee Jenkins wrote: I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. Interesting. But if that was the case, wouldn't restarting asterisk (#CLI restart now) clear the problem? Because on my system, it does not clear the problem, only a complete restart of the server seems to work. Well that and trying again some considerable time later which may or may not work then. -- Warm Regards, Lee ___ --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] Installed TDM02B - Problem when other end hangs up
Hi, Lee:
Lee Jenkins wrote:
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought (and posted incorrectly above) that I could dial into the
system from outside and reset the line, but that does not always work so
it really seems random to me. I think the only thing that definitely
resets the line so that it will work is rebooting the server unfortunately.
I'll bet you have exactly the same issue that I do.
What signalling are you using in your zapata.conf file?
-Stephen-
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Re: [asterisk-users] H.264 *Not Patented*
Snom's do 729 just fine 190,320,360 all have worked well for us On 27 Jan 2007, at 16:33, Lee Jenkins wrote: Although I wouldn't complain about a free G.729 codec, I have to be honest in saying that $10.00 isn't that great of an expense considering the better call quality you get. Does G.729 work by pushing up the compression, therefore moving from work from the Network to the CPU ? Across my LAN, I'd probably be able to handle *fewer*, rather than more calls across my * exchange if this was the case. If it's cleverer that this, I think I'll have to speculate a few dollars, assuming my Snoms can talk in G.729. cheers -a -- Regards, Andy Davidson http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Cisco has quite a bit of material on this subject (why should they special? I'll bet their telephony stuff is plagued by this problem too). This document describes different kinds of supervisory disconnect signalling: Understanding FXO Disconnect Problem http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml This one provides a good summary of signalling: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml#Topic3A I think the problem comes down to knowing what kind of signalling the telco is using (if it is using any at all) and then configuring the TDM accordingly. I wonder if anyone has had any luck getting that kind of information from any ILEC. -Stephen- ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. Interesting. But if that was the case, wouldn't restarting asterisk (#CLI restart now) clear the problem? Because on my system, it does not clear the problem, only a complete restart of the server seems to work. Well that and trying again some considerable time later which may or may not work then. That's because I assumed (based on what you said in previous posts) that the card thought the line was clear, but if your experience is anything like mine, the card leaves the channel open because it doesn't detect the disconnect signal from the CO. If the card still thinks the line is in use too, it will refuse access to the channel. That's why restarting Asterisk doesn't help. I'm getting a test dongle and a multimeter to see if I can observe a disconnect supervision signal on one of my lines. I'll let you know what I find. It will take me a few minutes. -Stephen- ___ --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] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Stephen Bosch wrote: Lee Jenkins wrote: I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. Interesting. But if that was the case, wouldn't restarting asterisk (#CLI restart now) clear the problem? Because on my system, it does not clear the problem, only a complete restart of the server seems to work. Well that and trying again some considerable time later which may or may not work then. Asterisk does not terminate active calls when doing a reload. I assume Asterisk thinks there is an active call on that port. ___ --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] Installed TDM02B - Problem when other end hangs up
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought (and posted incorrectly above) that I could dial into the
system from outside and reset the line, but that does not always work so
it really seems random to me. I think the only thing that definitely
resets the line so that it will work is rebooting the server unfortunately.
I'll bet you have exactly the same issue that I do.
What signalling are you using in your zapata.conf file?
fxs_ks
--
Warm Regards,
Lee
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[asterisk-users] Asterisk, VoIP and Linux Blog.
Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I would help) I can create categories for english talking people. To write a post, the only thing you have to do is register yourself, every article has to be aproved by a moderator, if it's well written, there will be no problem. I hope you like it. Regards. -- Facundo Ameal. famealatgmaildotcom http://fameal.blogdns.org Linux User #395088 Share your knowledge, use free software. ___ --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] Installed TDM02B - Problem when other end hangs up
Eric ManxPower Wieling wrote: Lee Jenkins wrote: Stephen Bosch wrote: Lee Jenkins wrote: I forgot to mention that the one thing that seems to be consistent is that I can get the zap line to reset and dialout again correctly by calling into the system on that zap line, dialing and extension and allowing the extension to hangup on the caller first. Then it will dial out again. Odd. Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. Interesting. But if that was the case, wouldn't restarting asterisk (#CLI restart now) clear the problem? Because on my system, it does not clear the problem, only a complete restart of the server seems to work. Well that and trying again some considerable time later which may or may not work then. Asterisk does not terminate active calls when doing a reload. I assume Asterisk thinks there is an active call on that port. ___ Hi Eric, I did a restart, not a reload. Actually I've tried restarting the server and where I originally thought it did reset, only randomly will restarting the whole server actually work. -- Warm Regards, Lee ___ --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 + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart [EMAIL PROTECTED] wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal [EMAIL PROTECTED] wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Installed TDM02B - Problem when other end hangs up
Stephen Bosch wrote: Lee Jenkins wrote: Well, not really -- when the extension hangs up, Asterisk knows the channel has been abandoned and clears it. When the remote party hangs up first, the card doesn't tell Asterisk that the channel is clear (because it doesn't know the caller has hung up), so Asterisk has no idea and presumes it is unavailable. Interesting. But if that was the case, wouldn't restarting asterisk (#CLI restart now) clear the problem? Because on my system, it does not clear the problem, only a complete restart of the server seems to work. Well that and trying again some considerable time later which may or may not work then. That's because I assumed (based on what you said in previous posts) that the card thought the line was clear, but if your experience is anything like mine, the card leaves the channel open because it doesn't detect the disconnect signal from the CO. If the card still thinks the line is in use too, it will refuse access to the channel. That's why restarting Asterisk doesn't help. I'm getting a test dongle and a multimeter to see if I can observe a disconnect supervision signal on one of my lines. I'll let you know what I find. It will take me a few minutes. Okay. Here's what I can tell you. Scenario: On hook state: line voltage is 52 VDC. Phone A calls Phone B Off hook state: line voltage is 7.5 VDC. Phone B answers Phone A and Phone B yak Phone B hangs up, but Phone A stays on the line After ~65 seconds (give or take a few milliseconds), the CO drops the voltage to 1.5 volts (from 7.5) for roughly half a second, then the voltage returns to 7.5. No polarity reversal (unless the reversal is so quick that my digital multimeter doesn't have time to display it, which is also a possibility; what I suppose really need is an analog meter... how ironic). So, my telco is doing some kind of disconnect signalling. The question is whether the TDM card can detect it. Either way, 65 seconds is a bloody eternity in the phone world. That's 65 seconds during which the line can't be used... Am I the only one who thinks that's insane? I'd be interested in some feedback from other users; how long does it normally take to get a disconnect signal in your area? The repair guy I talked to said that it was pretty much an impossibility that I would ever get them to reduce the delay and to forget about it immediately. I'm going to do some testing later today to see if that 65 second delay is consistent or totally random. -Stephen- ___ --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] NTL Hangup
On Friday 26 January 2007 23:40, Leo Ann Boon wrote: Kyle Gordon wrote: fxsks=1 #X100P Is your line truly a kwelstart line? try fxsls SNIP busydetect=yes You may need to add these 2 values to help the busydetect busycount=3 busypattern=375,375 busypattern tells asterisk how your busy tone sounds like, in UK it should be 400Hz 0.375s ON and 0.375s OFF. The busycount tells asterisk how many consecutive cycles it must detect before dropping the line. You'll have to determine the best value for your setup, by trial and error. Too low - you might get premature hangup, too high - you'll have to wait for a long time for the line to hangup. A value of 3 will cause Asterisk to hang up in about 2.1s. SNIP switchtype=national This is not needed for analog lines. signalling=fxs_ks Change to fxs_ls to match zaptel.conf Hi Leo, That appears to have done the trick. fxs_ls does seem to detect it hanging up more reliably. I don't know what the difference is, but it works :-) If there's any change, I'll be sure to let you know :-p Many thanks, Kyle -- Kyle Gordon [EMAIL PROTECTED] http://lodge.glasgownet.com pgpLlMUhplz8z.pgp Description: 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
[asterisk-users] Re: Cordless SIP Phones
-Original Message- John Marvin, Thank you very much. The CYT35 utility worked like a charm, though I feel a bit like a criminal. Not at all intuitive to set up, but the VTech 8100-2 is performing marvelously with my asterisk setup. I just got my grandstream budgetones in the mail, also, which worked out of the box with little configuration. So I've decided to keep all three phones. Thanks again. Ed ___ --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] RE: Realtime Voicemail Password Change Not Working
I was able to update the password through the dialplan with this:
exten = ,1,MYSQL(Connect connid 127.0.0.1 pbx pbx pbxdb)
exten = ,2,MYSQL(Query resultid ${connid} UPDATE\ voicemail\ SET\
password=\ where\ mailbox=52007)
exten = ,3,MYSQL(Clear ${resultid})
exten = ,4,MYSQL(Disconnect ${connid})
exten = ,5,Hangup
Finaly I got an update statement in the mysql log:
12 Query UPDATE voicemail SET password= where mailbox=52007
So these results suggest that mysql, voicemail table, and the res_mysql
adddon are working fine. It suggests that app_voicemail is not passing
the
update statement to the res_mysql driver.
This was a clean install, nothing out of the ordinary.
I would second the other posters suggestion: use Realtime update (show
application realtime update) since it uses the actual realtime setup. The
MySQL command shown above uses a new connection that you specify so is not
such a good test.
Cameron
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Re: [asterisk-users] Installed TDM02B - Problem when other end hangs up
Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. When the problem occurs, calling that ZAP line from outside seems to reset it as well. I'm sure it's something obvious I've overlooked, but I'd appreciate any pointers. I've been working on getting this card to work correctly just about all day and while I'm certainly no expert, I just not sure it's a problem with asterisk holding the line open as others have suggested. My reasoning follows and please let me know if I'm off base here. 1. Attempt to dial out on zap line. CLI says it's dialing but nothing happens and after a while the phone company comes back with if you'd like to make a call, please hangup and try again followed by congestion. 2. Repeat attempting to dial out as many times as I like and get the same result maybe every time and then out of the blue, bang! it goes through usually just once. Very random and rarely can I make two calls in a row. If one does in fact go through, the zap line is screwed up for a while after that regardless if I hang up or the other part hangs up. 3. During the periods when I cannot dial out of the zap line, I can actually dial into that same line from my cell phone any time I like. Call after call, it goes through when dialing into the system. Then I will attempt to dial out and same problem occurs: the zap line appears to dial out but nothing actually happens. My point is that if Asterisk really did have the line stuck open, I wouldn't be able to dial in on that same line, right? But I can, time after time, but dialing out is almost always a no go. Am I off base? -- Warm Regards, Lee ___ --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] NTL Hangup
Kyle Gordon wrote: snip Hi Leo, That appears to have done the trick. fxs_ls does seem to detect it hanging up more reliably. I don't know what the difference is, but it works :-) If there's any change, I'll be sure to let you know :-p No problemo. Glad to know it worked for you. Like Tzafrir said, this is one of the less documented aspect of asterisk. Leo ___ --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] Installed TDM02B - Problem when other end hangs up
Hi, Lee: Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. When the problem occurs, calling that ZAP line from outside seems to reset it as well. I'm sure it's something obvious I've overlooked, but I'd appreciate any pointers. I've been working on getting this card to work correctly just about all day and while I'm certainly no expert, I just not sure it's a problem with asterisk holding the line open as others have suggested. My reasoning follows and please let me know if I'm off base here. 1. Attempt to dial out on zap line. CLI says it's dialing but nothing happens and after a while the phone company comes back with if you'd like to make a call, please hangup and try again followed by congestion. What exactly do you mean by followed by congestion? Are you sure that the line is already clear when you dial out? (I'll bet you five bucks it's not.) Can you confirm that the TDM card is actually dialing? Try listening in on the call by plugging in an extension between the TDM and the incoming line. Can you confirm a dial tone before the TDM card dials? You can pick up the extension before the TDM card goes off-hook and listen for a dial tone. The card won't care if the line is already off-hook; it should just dial if it detects a dial tone. Monitoring the process with an dumb extension not connect to Asterisk would be very illuminating. 2. Repeat attempting to dial out as many times as I like and get the same result maybe every time and then out of the blue, bang! it goes through usually just once. Very random and rarely can I make two calls in a row. If one does in fact go through, the zap line is screwed up for a while after that regardless if I hang up or the other part hangs up. IF the TDM card is actually dialing when it hears a dial tone, and the call doesn't go through, then it's possible your DTMF duration is too short (that is, the duration of the tone for a specific digit is so short the switch doesn't recognize it). Some crappy phone sets don't generate DTMF if the line polarity is backwards, which happens more often than you might think (lots of wire installations are done incorrectly). This isn't FCC compliant, however, as phone devices aren't supposed to care what the polarity is. I'm pretty sure the TDM doesn't either. Can you show us your zaptel.conf and zapata.conf? 3. During the periods when I cannot dial out of the zap line, I can actually dial into that same line from my cell phone any time I like. Call after call, it goes through when dialing into the system. Then I will attempt to dial out and same problem occurs: the zap line appears to dial out but nothing actually happens. Appears and does are very different, as I'm sure you'll agree. You need to find out if the card is *actually* dialing when you try and call out. What do you mean by goes through -- does Asterisk answer the call and pass it to an extension, IVR, or voice mail? Or does the caller hear ringing while nothing happens on the Asterisk end? I still think the call clearing issue is at least part of your problem, but the good news is that outgoing call issues on an available line are easier to fix. My point is that if Asterisk really did have the line stuck open, I wouldn't be able to dial in on that same line, right? But I can, time after time, but dialing out is almost always a no go. Let's see those configs, and relevant debug log output if you can arrange it (asterisk -vvvdc). Cheers, -Stephen- ___ --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] TDM Cards or PSTNVOIP Gateways?
OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a w in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping the line open (which I doubted based on the fact that I could call into that line anytime) but instead that the card was not coming on line fast enough and Asterisk was just pushing part of the phone number to dial to the card. My thanks to Rick Neubaurer who suggested the fix. -- Warm Regards, Lee ___ --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] TDM Cards or PSTNVOIP Gateways?
Lee Jenkins wrote: OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a w in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping the line open (which I doubted based on the fact that I could call into that line anytime) but instead that the card was not coming on line fast enough and Asterisk was just pushing part of the phone number to dial to the card. My thanks to Rick Neubaurer who suggested the fix. I completely sent the wrong header in this post. Sorry folks, this was mean the for the thread: Installed TDM02B - Problem when other end hangs up. I was going to ask this THIS thread what others thought about the Gateway products like the Grandstream 4104 that I say posted earlier this week. Anyone else have any thoughts on these products? Personally, I'd like to just purchase TDM cards if I can to support digium, but if the Gateways are easy to install and provide a way to offload processing from the server then maybe I'll buy one and tinker with it. Just curious if anyone else had any input on these devices. Again, sorry about the wrong initial post. -- Warm Regards, Lee ___ --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] Installed TDM02B - Problem when other end hangs up
Stephen Bosch wrote: Hi, Lee: Lee Jenkins wrote: Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing happens and I get the telco's message if you'd like to make a call, hang up... after a few seconds. If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. When the problem occurs, calling that ZAP line from outside seems to reset it as well. I'm sure it's something obvious I've overlooked, but I'd appreciate any pointers. I've been working on getting this card to work correctly just about all day and while I'm certainly no expert, I just not sure it's a problem with asterisk holding the line open as others have suggested. My reasoning follows and please let me know if I'm off base here. 1. Attempt to dial out on zap line. CLI says it's dialing but nothing happens and after a while the phone company comes back with if you'd like to make a call, please hangup and try again followed by congestion. What exactly do you mean by followed by congestion? Are you sure that the line is already clear when you dial out? (I'll bet you five bucks it's not.) Can you confirm that the TDM card is actually dialing? Try listening in on the call by plugging in an extension between the TDM and the incoming line. Can you confirm a dial tone before the TDM card dials? You can pick up the extension before the TDM card goes off-hook and listen for a dial tone. The card won't care if the line is already off-hook; it should just dial if it detects a dial tone. Monitoring the process with an dumb extension not connect to Asterisk would be very illuminating. 2. Repeat attempting to dial out as many times as I like and get the same result maybe every time and then out of the blue, bang! it goes through usually just once. Very random and rarely can I make two calls in a row. If one does in fact go through, the zap line is screwed up for a while after that regardless if I hang up or the other part hangs up. IF the TDM card is actually dialing when it hears a dial tone, and the call doesn't go through, then it's possible your DTMF duration is too short (that is, the duration of the tone for a specific digit is so short the switch doesn't recognize it). Some crappy phone sets don't generate DTMF if the line polarity is backwards, which happens more often than you might think (lots of wire installations are done incorrectly). This isn't FCC compliant, however, as phone devices aren't supposed to care what the polarity is. I'm pretty sure the TDM doesn't either. Can you show us your zaptel.conf and zapata.conf? 3. During the periods when I cannot dial out of the zap line, I can actually dial into that same line from my cell phone any time I like. Call after call, it goes through when dialing into the system. Then I will attempt to dial out and same problem occurs: the zap line appears to dial out but nothing actually happens. Appears and does are very different, as I'm sure you'll agree. You need to find out if the card is *actually* dialing when you try and call out. What do you mean by goes through -- does Asterisk answer the call and pass it to an extension, IVR, or voice mail? Or does the caller hear ringing while nothing happens on the Asterisk end? I still think the call clearing issue is at least part of your problem, but the good news is that outgoing call issues on an available line are easier to fix. My point is that if Asterisk really did have the line stuck open, I wouldn't be able to dial in on that same line, right? But I can, time after time, but dialing out is almost always a no go. Let's see those configs, and relevant debug log output if you can arrange it (asterisk -vvvdc). Cheers, Thanks Stephen, I have I posted the fix in another post to this thread. But quoted below for you: OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a w in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping the line open (which I doubted based on the fact that I could call into that line anytime) but instead that the card was not coming on line fast enough and Asterisk was just pushing part of the phone number to
Re: [asterisk-users] Installed TDM02B - Problem when other end hangsup
Yuan LIU wrote: From: /Lee Jenkins [EMAIL PROTECTED]/ [...] If I call out to a party on that Zap line and hangup first, I do not experience that problem. It looks like Asterisk is not getting the termination signal from the telco (Verizon) when the other party hangs up first. Running show channels at any time after the call is disconnected (by either party) shows 0 active calls/channels. [...] I've been working on getting this card to work correctly just about all day and while I'm certainly no expert, I just not sure it's a problem with asterisk holding the line open as others have suggested. My reasoning follows and please let me know if I'm off base here. 1. Attempt to dial out on zap line. CLI says it's dialing but nothing happens and after a while the phone company comes back with if you'd like to make a call, please hangup and try again followed by congestion. 2. Repeat attempting to dial out as many times as I like and get the same result maybe every time and then out of the blue, bang! it goes through usually just once. Very random and rarely can I make two calls in a row. If one does in fact go through, the zap line is screwed up for a while after that regardless if I hang up or the other part hangs up. 3. During the periods when I cannot dial out of the zap line, I can actually dial into that same line from my cell phone any time I like. Call after call, it goes through when dialing into the system. Then I will attempt to dial out and same problem occurs: the zap line appears to dial out but nothing actually happens. My point is that if Asterisk really did have the line stuck open, I wouldn't be able to dial in on that same line, right? But I can, time after time, but dialing out is almost always a no go. This must be correct. Additionally, you cannot send an off-hook signal to CO if the line is already open, thus you will not be hearing CO announcing if you'd like to make a call. Telco will also drop your line dead if you are off-hook idle for too long. So I'd suspect that Asterisk isn't dialing. However, this would not explain why this only happens when the other side hang up first. May be try insert ChanIsReady() before dial, just to test out? There are some electrical tests you can do to determine line status when Asterisk got stuck. Line DC voltage is a sure indication of whether your FXO is on-hook or off-hook. Yuan Liu Hi Yuan, Thanks for chiming in. I accidentally posted the fix to a wrong thread above. Been a very long day ;) Here is what I posted: OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a w in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping the line open (which I doubted based on the fact that I could call into that line anytime) but instead that the card was not coming on line fast enough and Asterisk was just pushing part of the phone number to dial to the card. My thanks to Rick Neubaurer who suggested the fix. -- Warm Regards, Lee ___ --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 + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi, I'm not sure, but I experienced it before with our Nortel Meridian I MFC/R2. When set to both zero(0), calls drop once answered. I tried to vary its values until I finally got it stabled. I'd been in the Datacoms/Telecoms for 16 years now, only with Asterisk I experienced beyond technical theory (out of the book). But bottom line is, it works. Magic ! Angel. Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and
[asterisk-users] disconnect clear time -- calling party control and TDM-400
Hi: Is there any way to adjust the detection threshold for kewlstart signalling on the TDM-400 cards? Example: The telco provides a 100 ms open loop or battery drop to indicate remote party hangup. If the zaptel driver expects to see a 350 ms drop, it will never detect the hangup and sit on the line. Many PBXs let you adjust this number. Any ideas, anyone? Thanks, -Stephen- ___ --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 + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
I'll try it during weekend then. Thanks for the help. I appreciate it. On 1/29/07, Angel Heart [EMAIL PROTECTED] wrote: Hi, I'm not sure, but I experienced it before with our Nortel Meridian I MFC/R2. When set to both zero(0), calls drop once answered. I tried to vary its values until I finally got it stabled. I'd been in the Datacoms/Telecoms for 16 years now, only with Asterisk I experienced beyond technical theory (out of the book). But bottom line is, it works. Magic ! Angel. *Facundo Ameal [EMAIL PROTECTED]* wrote: Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088
Re: [asterisk-users] T1 Wire Level Tapping
Shane Spencer wrote: I am very interested in the DACs capabilities of Digium cards, there is no information anywhere on this. I could always do pri bridging via libpri like you suggest however. But having hardware handle the bridging onboard a single PCI card would help reduce my server requirements for a final product, as long as I can spy on active channels somehow. I don't think its going to work that way, I wil test out libpri for a bit. Pardon if I'm wrong, I don't think the DACS mode is really applicable if you're trying to monitor the channels. As I understand it, if you use DACs - the data will just flow between the 2 ports and not to the PCI bus. So logically, you won't be able to spy on the channels. Leo ___ --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] T1 Wire Level Tapping
I wanted to know if there was a peekaboo factor to it all. You can flow data under a glass window :) On 1/29/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am very interested in the DACs capabilities of Digium cards, there is no information anywhere on this. I could always do pri bridging via libpri like you suggest however. But having hardware handle the bridging onboard a single PCI card would help reduce my server requirements for a final product, as long as I can spy on active channels somehow. I don't think its going to work that way, I wil test out libpri for a bit. Pardon if I'm wrong, I don't think the DACS mode is really applicable if you're trying to monitor the channels. As I understand it, if you use DACs - the data will just flow between the 2 ports and not to the PCI bus. So logically, you won't be able to spy on the channels. Leo ___ --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] TDM 400P in the UK - doesn't see ringing calls hanging up before answer
Ed W wrote: Using a TDM400P in the UK nearly works fine, but I have a last remaining problem in that if the incoming is ringing and then the caller hangs up, asterisk takes another couple of rings before it spots the hangup. This is annoying in that I sometimes get phantom calls late at night (possibly due to call waiting or the exchange doing a half ring to see if we are live). Also I get phantom calls on either the voicemail or when I answer there is just dial-tone because the caller hungup before the call was answered I have fiddled with a number of settings relating to polarity reversal because I believe that might be relevant to BT's implementation, but it's not made any difference from the default config. Any suggestions on how to fix this from UK users? I have tried most of the suggestions in the voip wiki to no effect (haven't tried calling BT and asking them to change any settings yet) Did you end up calling BT? I'm curious. -Stephen- ___ --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] T1 Wire Level Tapping
Shane Spencer wrote: I wanted to know if there was a peekaboo factor to it all. You can flow data under a glass window :) Well - you can always use a logic probe :). Bridging does add a little latency to the whole thing. Why don't you consider a passive tap solution like the hi-z OpenPRI card from voicetronix? It doesn't cost much more than a solution based on digium hardware. Leo ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
you got that while doing SIP/ZAP and parking? On 1/29/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 29 Jan 2007, Steve Davies wrote: I failed to notice that it was included in 51363 - I just checked, and that change is indeed already in. Sorry, my mistake. I generally do not change the -march setting, so I am probably using an i386 default. I get segfaults with the VIA C3 and C7 chips (on CN1000 and other EPIA boards) with I leave it as the defaults. I need the -i586 option. -i686 seems the be the default in the makefile. I understand it's to do with the MMX instructions used in some of the codecs... Gordon Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Cisco PRI gateway with MGCP control
Hello, Anyone managed to control a Cisco voice gateway (2,811 in my case) using MGCP? I cannot make the PRI going on-line (while with SIP I can). If you ask why I want to use MGCP and not SIP: it is because Cisco uses different Q.sig signalling when you manage it with different protocols, and I need the other Q.sig... Thanks, __Yehavi: ___ --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] detecting avaya busy tone
n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Installed TDM02B - Problem when other end hangsup
From: Lee Jenkins [EMAIL PROTECTED] Hi Yuan, Thanks for chiming in. I accidentally posted the fix to a wrong thread above. Been a very long day ;) Here is what I posted: OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Probably the card is a little too fast for the line:-) Glad to see a problem solved. My telco would often give me the call cannot be completed as dialed, please try again when incomplete digits are pressed, instead of if you'd like to place a call. That makes it eaiser to diagnose. Yuan Liu Placing a w in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping the line open (which I doubted based on the fact that I could call into that line anytime) but instead that the card was not coming on line fast enough and Asterisk was just pushing part of the phone number to dial to the card. My thanks to Rick Neubaurer who suggested the fix. -- Warm Regards, Lee ___ --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] detecting avaya busy tone
What avaya system is this, if the avaya is configured on the ports to use a 2500 set, then it should do CPC and should work as is. On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Heartbeat on Digium T1 PCI cards?
Shane, are you trying to say that the PRI was actualy down (the D channel was NOT up) for the time that ATT is billing you? On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote: Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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 ___ --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 ___ --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] Timeout in IAX vs SIP
When Asterisk dials an IAX destination with no registration, it very quickly comes to the conclusion that it can't make the call -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] [Jan 29 21:43:15] NOTICE[1957]: chan_iax2.c:2686 __auto_congest: Auto-congesting call due to slow response -- IAX2/216.207.245.8:4569-1 is circuit-busy -- Hungup 'IAX2/216.207.245.8:4569-1' == Everyone is busy/congested at this time (1:0/1/0) But if Asterisk Dials a SIP destination it doesn't have a registration, it waits for a very long time before giving up. What is the difference? Does IAX use TCP instead of UDP? Is there some way to change timeout value in SIP attempt so it gives up in a reasonable time? Yuan Liu ___ --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] detecting avaya busy tone
This is a G3. And I'm not the avaya operator. What do you mean with 2500 set and CPC? On 1/29/07, C F [EMAIL PROTECTED] wrote: What avaya system is this, if the avaya is configured on the ports to use a 2500 set, then it should do CPC and should work as is. On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Heartbeat on Digium T1 PCI cards?
It was either down or asterisk was frozen. Either way a heartbeat could fix that. On 1/29/07, C F [EMAIL PROTECTED] wrote: Shane, are you trying to say that the PRI was actualy down (the D channel was NOT up) for the time that ATT is billing you? On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote: Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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 ___ --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 ___ --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
[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --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 -- Sergio de los Santos ssantos @ hispasec.com Hispasec Sistemas S.L 902 161 025 29590 Málaga ___ --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
