RE: [asterisk-users] Running Multiple Instances of Asterisk
Thirdlane PBX Manager multi-instance can be used to manage/configure multiple instances of Asterisk. If you have any questions please contact me at [EMAIL PROTECTED] Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, September 25, 2006 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk I don't see a problem here. Using includes you dedicate every company their own directory of configs. Macros are eithere system wide, or each comapny can create their own. I don't see why this is any harder than mutilple instances of asterisk. On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Doug ___ --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: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Well, it would seem to me that with a little attention to processor affinity, you could run your Asterisk and DBMS code on one processor, and let the other one handle the device interrupts; ie: that sounds to me like a feature, rather than a bug... Ok, and if I have two dual core processors? It doesn't sound like very useful feature to me (in some situations). -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Running Multiple Instances of Asterisk
We offer a management GUI for both options - multi-tenant (multiple companies within the same instance of Asterisk) or multi-instance (multiple instances of Asterisk). Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Monday, September 25, 2006 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Douglas Garstang wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Have you tried running asterisk in a chroot environment? It can do what you want. The only catch you'll have to specify the bindaddr for SIP. And, it works with IP aliases, so you can host multiple sessions on one NIC. Cheers. ___ --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: Dual core
For what we do with Asterisk(lots of meetme and Zap - IAX2) It does spread the load across both cores. In our initial comparisons for equal call traffic, the P4-D had half or the average loadavg for a 6 hour time period of the P4 of the same speed. MATT--- Hi Matt! Thank you for information's. Can you please tell me have you made any special adjustments or steps in Asterisk install or configuration to achieve this? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
Tomislav Parčina ha scritto: For what we do with Asterisk(lots of meetme and Zap - IAX2) It does spread the load across both cores. In our initial comparisons for equal call traffic, the P4-D had half or the average loadavg for a 6 hour time period of the P4 of the same speed. MATT--- Hi Matt! Thank you for information's. Can you please tell me have you made any special adjustments or steps in Asterisk install or configuration to achieve this? The dual core solution (even for AMD or INTEL) is a good solution when the server that run asterisk need more cpu to do something. There is no need to configure, you add the CPU and the kernel (if SMP) does the rest. Only if you want to bound a single CPU to a single interrupt you can change the affinity of the CPU. Or If you use irqbalance this affinity change is done by the irqbalance daemon. In a real context (with asterisk) is better to NOT use irqbalance and set affinity for each voip board/irq on the system. Bye. 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
RE: [asterisk-users] trixbox t38 pass through
T38 passthrough doesnt seem to work in trunk at the moment. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: 26 September 2006 10:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] trixbox t38 pass through Most people here don't give a damn about anything but their own personal problems. Sounds like you are probably one of them, Thanks for noticing! ___ --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] Set hint status from dialplan?
Is it possible to manually set the hint status of a virtual extension via the dialplan?I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn on a light for night mode.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstatefor more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activationsexten = 799,hint,DS/mmgcexten = 799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)exten = 799,n,Playback(beep)exten = 799,n,Hangup
[macro-open-close]exten = s,1,DBGet(nightmode=nightmode/${ARG1})exten = s,n,NoOp(ARG1 ${ARG1} ARG2 ${ARG2} ARG3 ${ARG3})exten = s,n,Set(OpenFile=${ARG2})exten = s,n,Set(CloseFile=${ARG3})
exten = s,n,NoOp(Close file ${CloseFile}. Open file ${OpenFile}.)exten = s,n,GotoIf(${nightmode}=1?s,Open:s,Close)exten = s,n(Open),DBPut(nightmode/${ARG1}=0)exten = s,n,Devstate(${ARG1},0)
exten = s,n,Playback(${OpenFile})exten = s,n,Goto(Return)exten = s,n(Close),DBPut(nightmode/${ARG1}=1)exten = s,n,Devstate(${ARG1},2)exten = s,n,Playback(${CloseFile})exten = s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it turns night mode on and announces that, another press and it turns night mode off and announces that..
-- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own...
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Re: [asterisk-users] Asterisk with cisco 7935
Just wondering if anyone has had any luck getting the cisco 7935 workingwith asterisk and if so, what is the best way to go about it?on the The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP firmware that was releasedthis month. I don't know if it works any better. -- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... ___ --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] Problem with zaptel 1.4b2 and X101P Wildcard
On Mon, Sep 25, 2006 at 09:27:20PM +0200, Morten Isaksen wrote: On 9/25/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] zaptel-1.4.0-beta1]# ztcfg -vvv Notice: Configuration file is /etc/zaptel.conf line 235: Unable to read Zaptel version information. Zaptel Version: $êþP¦0 Echo Canceller: Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) What exactly is channel 1? Maybe you got the wrong number? cat /proc/zaptel/* [EMAIL PROTECTED] zaptel-1.4.0-beta1]# service zaptel start Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) [FAILED] [EMAIL PROTECTED] zaptel-1.4.0-beta1]# cat /proc/zaptel/* Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 The same configuration works perfect with zaptel 1.2.1 Next thing to do, I guess, is to run: strace ztcfg to see which device exactly is accessed . Though /dev/zap/ctl is the usual suspect. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The best way to track no-audio calls
Hi,We're breaking our teeth on a 1.2.10 - 0.3.0-PRE-1s bristuffed Asterisk which sometimes refuses to pass audio through.The trouble is we cannot a practical way to identify no-audio calls without involving end users as we don't want to bother them with that. What could be the best method to identify no-audio calls ?Here is what I was thinking about.- The setup is:Public Network -- Asterisk with Junghanns QuadBRI Voice Lan --- Snom 320 IP hardphones - The symptoms are :Sometimes, during normal call (good audio quality), audio is 2-way cut (Snom sends BYE requests).Sometimes, when a user answers incoming call, he doesn't ear anything. So he hangs up and next call from the same caller, a couple of seconds later, is perfect. Nothing abnormal in log files (from my point of view anyway though I know this will soon be proven to be wrong, when we have found the reason why this happens).- As this behaviour touches every phone and every ISDN channel, 1st step is to focus on a small subset of randomly chosen users. - For focused users, record everything : use tcpdump everything coming from hardphone's IP address (as every call passes through the server), use MixMonitor to record every call as MixMonitor acts on a different level (I've never used MixMonitor in my life and I'm honnestly wondering if recording on both tcpdump and MixMonitor is of any use) increase log levels for focused users.What do you think of these steps ?Regards ___ --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 with cisco 7935
On 16/09/2006, at 6:43 AM, Doug Lytle wrote: Miles Scruggs wrote: What did you end up using for conference stations? Polycom IP 501s Doug We are using the Polycom IP4000 conference phones, same software as the IP501/601 etc..etc.. series. Works very well. Nathan. ___ --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] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma
Hi, I have an Asterisk box with a Sangoma a102 making bridge between an old legacy PBX and a PRI line. Our customer told us he could not call some numbers so we investigated and found he was right: each time we tried to call some numbers the other party hang up. It could be be the other party PBX so we acted on zapata.conf and finally we found the right parameters configuration and we could call those numbersbut now it is not possible to receive inbound calls. Is something related to http://bugs.digium.com/print_bug_page.php?bug_id=7511? Why I cannot receive inbound calls?? Is there anybody experienced situation like this? TIA Giorgio Incantalupo (zapata.conf follows) ; this channels are connected to PRI context = outbound_calls group = 1 immediate = no internationalprefix = 00 language = us nationalprefix = 0 pridialplan = unknown prilocaldialplan = unknown priindication = inband resetinterval = never signalling = pri_cpe switchtype = national usecallerid = yes callerid = asreceived overlapdial=no relaxdtmf=yes usedistinctiveringdetection=yes channel = 1-15,17-31 ; this channels are connected to legacy PBX context = legacy_zap group = 2 immediate = no ;internationalprefix = 00 language = us ;nationalprefix = 0 ;pridialplan = national priindication = inband resetinterval = never signalling = pri_net switchtype = euroisdn ;usecallerid = yes callerid = asreceived relaxdtmf=yes overlapdial=yes ; YES is mandatory here usedistinctiveringdetection=yes channel = 32-46,48-62 ___ --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] Changing the recording dir of MeetMe recordings.
Hi.
I'm using the default recording file name for meetme recordings of
meetme-conf-rec-${CONFNO}-${UNIQUEID}
The recordings are created in /var/lib/asterisk/sounds.
I want to change this direcrory to /var/lib/asterisk/sounds/storagedrive.
Settig the ${MEETME_RECORDINGFILE} variable doesn't help. It uses the
correct directory name but then just create the file as
meetme-conf-rec--.wav as it doesn't know the values of CONFNO and
UNIQUEID at that stage.
Can somebody please help me to achieve this directory change.
Thank you very much.
Regards.
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[asterisk-users] Asterisk and UMTS phones
Hello everyone, did anyone attempt to use Asterisk with UMTS third generation mobiles? I'd like to route some calls to an UMTS phones, but I don't know if this can be done/is likely to be done/will be available someday/is impossible. Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Segmentation fault on Asterisk startup:res_config_mysql.so problem?
Did you do a make make install for add-ons BEFORE doing so for asterisk? If so try asterisk first and when all is installed install add-ons. -- I tried a make clean make make install for asterisk and then for asterisk-addons but am still getting the segmentation fault on asterisk startup. rm res_config_mysql.so allows Asterisk to start. Any other suggestions appreciated. 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
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
I also think you get what you pay for and I don't use hfc based isdn cards in production any more. Having said that, a small home installation isn't quite the same as a 30 user office environment. My home-pbx for example is quite happy reloading asterisk+zaphfc every night. Of course not something I'd accept in a production environment, but that's probably not what HFC-s cards are aimed at either, right? Imho, the bri cards based on hfc chipsets work very well.. I too have some problems with bristuff, but I don't think the card is to be blamed here.. The mISDN way looks like a better way, the BRI cards of digium also use it.. And they're not cheap.. So I guess it's time (for me at least) to reconsider the setup and try out the mISDN-way.. It's no problem to use mISDN with the quadbri of junghanns or beronet, they're all hfc-based.. cheers ___ --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: asterisk to cell phone network
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1/2/4 simslot pci card: http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. Where can I see the prices of this cards? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] asterisk 1.4 branch and chan-capi-0.7.0
Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? System: Fedora Core 4 with Kernel 2.6.17-1.2142_FC4 AVM Fritz Card is present and fcpci running and up isdn4k-utils and isdn4k-utils-devel installed capi4hylafax installed make in chan_capi source said: gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:146: warning: type defaults to 'int' in declaration of 'STANDARD_LOCAL_USER' chan_capi.c:146: warning: data definition has no type or storage class chan_capi.c:147: warning: type defaults to 'int' in declaration of 'LOCAL_USER_DECL' chan_capi.c:147: warning: data definition has no type or storage class chan_capi.c: In function 'capi_new': chan_capi.c:2078: error: 'struct ast_channel' has no member named 'type' chan_capi.c: In function 'pbx_capicommand_exec': chan_capi.c:4582: warning: implicit declaration of function 'LOCAL_USER_ADD' chan_capi.c:4597: warning: implicit declaration of function 'LOCAL_USER_REMOVE' chan_capi.c: At top level: chan_capi.c:5244: error: unknown field 'send_digit' specified in initializer chan_capi.c:5244: warning: initialization from incompatible pointer type make: *** [chan_capi.o] Error 1 Thanks for any hints and ideas Guido ___ --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 1.4 branch and chan-capi-0.7.0
-Ursprüngliche Nachricht- Von: Armin Schindler [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 26. September 2006 13:37 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0 On Tue, 26 Sep 2006, Guido Hecken wrote: Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? chan-capi was not ported to Asterisk 1.4 yet. See bug http://bugs.melware.net/mantis/view.php?id=20 Armin Armin, thanks for the info. Are there any plans on porting it to 1.4 and if yes, is there an approximate release date? Regards, Guido ___ --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] core dump with 1.2.7.1 and chan-capi-cm 0.6.5
On Tue, 26 Sep 2006, Klaus Darilion wrote:
Hi Armin!
My Asterisk crashes once a day. The backtrace is:
Reading symbols from /usr/lib/asterisk/modules/func_enum.so...done.
Loaded symbols for /usr/lib/asterisk/modules/func_enum.so
Reading symbols from /usr/lib/asterisk/modules/func_uri.so...done.
Loaded symbols for /usr/lib/asterisk/modules/func_uri.so
Reading symbols from /lib/libnss_dns.so.2...done.
Loaded symbols for /lib/libnss_dns.so.2
#0 capi_handle_dtmf_fax (c=0x81b5b80) at chan_capi.c:1923
1923if (p-faxhandled) {
(gdb) bt
# 0 capi_handle_dtmf_fax (c=0x81b5b80) at chan_capi.c:1923
# 1 0x405df41f in capi_handle_facility_indication (CMSG=0xbd9ff944,
PLCI=258, NCCI=258, i=0x818a520) at chan_capi.c:2671
# 2 0x405dcb66 in capi_handle_msg (CMSG=0xbd9ff944) at chan_capi.c:3505
# 3 0x405da146 in do_monitor (data=0x0) at chan_capi.c:4186
# 4 0x40024e51 in pthread_start_thread () from /lib/libpthread.so.0
# 5 0x401ec8aa in clone () from /lib/libc.so.6
(gdb) quit
Is this is a known problem fixed in chan_capi-cm 0.7?
Yes, this is fixed in 0.7. At least it didn't show up again ;-)
Armin
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asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)
T38 passthrough doesn't seem to work in trunk at the moment. that's true http://bugs.digium.com/view.php?id=7679 http://bugs.digium.com/view.php?id=7844 t.38 in asterisk 1.4 http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 --- Marek Cervenka === ___ --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] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma
Klaus, the problem is for incoming calls from telco. Thanks for the hints...I'll try asap. :) Giorgio Klaus Darilion wrote: Giorgio Incantalupo wrote: Hi, I have an Asterisk box with a Sangoma a102 making bridge between an old legacy PBX and a PRI line. Our customer told us he could not call some numbers so we investigated and found he was right: each time we tried to call some numbers the other party hang up. It could be be the other party PBX so we acted on zapata.conf and finally we found the right parameters configuration and we could call those numbersbut now it is not possible to receive inbound calls. Is something related to This is too less information to find your problem. What is the problem in receiving incoming calls? Activate PRI debugging set verbose 9 pri debug span 1 pri debug span 2 and watch the q931 signaling. Where is the problem with the incoming call: between PSTN and Asterisk or between Asterisk and the PBX? I guess you have problems with the Type of Number (pridialplan , prilocaldialplan ) regards klaus http://bugs.digium.com/print_bug_page.php?bug_id=7511? Why I cannot receive inbound calls?? Is there anybody experienced situation like this? TIA Giorgio Incantalupo (zapata.conf follows) ; this channels are connected to PRI context = outbound_calls group = 1 immediate = no internationalprefix = 00 language = us nationalprefix = 0 pridialplan = unknown prilocaldialplan = unknown priindication = inband resetinterval = never signalling = pri_cpe switchtype = national usecallerid = yes callerid = asreceived overlapdial=no relaxdtmf=yes usedistinctiveringdetection=yes channel = 1-15,17-31 ; this channels are connected to legacy PBX context = legacy_zap group = 2 immediate = no ;internationalprefix = 00 language = us ;nationalprefix = 0 ;pridialplan = national priindication = inband resetinterval = never signalling = pri_net switchtype = euroisdn ;usecallerid = yes callerid = asreceived relaxdtmf=yes overlapdial=yes ; YES is mandatory here usedistinctiveringdetection=yes channel = 32-46,48-62 ___ --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] Re: Dual core
For the Asterisk installation, no. For Linux, yes. I built a custom SMP kernel, which depending on your Linux distribution may or may not be necessary for you. MATT--- On 9/26/06, Tomislav Parčina [EMAIL PROTECTED] wrote: For what we do with Asterisk(lots of meetme and Zap - IAX2) It does spread the load across both cores. In our initial comparisons for equal call traffic, the P4-D had half or the average loadavg for a 6 hour time period of the P4 of the same speed. MATT--- Hi Matt! Thank you for information's. Can you please tell me have you made any special adjustments or steps in Asterisk install or configuration to achieve this? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] error 407 authenticate on INVITE
Hi I noticed that in the normal flow of invite message, there is a message 407 in between. UA1 -INVITE -- AST1 407-- --ACK --INVITE--- --Trying-- There is a problem if 407 exists between asterisks. UA1 UA2 | | AST1 -INVITE -- AST2 407-- --ACK (AST1 don't know how to response the message 407 and the call will drop) I want to remove such authentication between asterisk such that call can flow between them. Is it possible to do that? How? 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] Set hint status from dialplan?
Is it possible to manually set the hint status of a virtual extension via the dialplan? I have an extension that turns my night mode on and off. I would love to be able to manually set the hint to be able to turn on a light for night mode. This is not a real (channel) extension, so there should be no harm to setting this manually. If this could be done, I could see a lot of opportunities to set lights for info that is not channel related. Also, could it be possible to read the hint status from the dialplan to make a button have different functionality based on a channels status? Please advise. -- -- Steven ___ --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] How can I stop lost DNS from killing Asterisk?
Rich Adamson wrote: Eric ManxPower Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. you can also install a non-crashing DNS server ;D ___ --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] University dumps CISCO VoIP for Asterisk
Andrew Kohlsmith wrote: On Wednesday 20 September 2006 21:40, Douglas Garstang wrote: We stuck OpenSER in between the phones and Asterisk, and pointed our phones towards the OpenSER boxes for SIP registrations and subscriptions. When OpenSER received a REGISTER or SUBSCRIBE message, it would use the send() command to forward the messages onto each Asterisk server. By doing that, ALL of our Asterisk servers had a copy of all sip registrations and subscriptions. It seemed to work pretty well, but for unrelated reasons, we dropped that approach. Which approach do you use now? what were the unrelated reasons ? ___ --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] Asterisk 1.4 mohsuggest
I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of documentation isn't helping much. I have this in sip.conf: [3254101] type=friend ... mohsuggest=class1 [3254102] type=friend ... mohsuggest=class2 A call is bridged between the two extensions. When 3254102 puts 3254101 on hold, 3254101 hears moh class 'class2' which is correct. However, when 3254101 puts 3254102 on hold, the 3254102 hears the default music class. Why? Doug. ___ --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] ztcfg / X100P question
Tzafrir Cohen wrote: On Tue, Sep 26, 2006 at 08:45:49AM -0400, Michel Vaillancourt wrote: Tzafrir Cohen wrote: what's the contents of /etc/zaptel.conf ? pbx1:~# cat /etc/zaptel.conf # # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us # fxsks=1 # channels=1 This line is unnecessary. Just remove it. Oh, *Duh*. I am so used to setting up PRIs... Thanks. pbx1:~# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) ... Now I need to figure out why its not seeing the card here. However, at least the Bizzare Error(tm) is out of the way. Thanks again. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --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] ztcfg / X100P question
Tzafrir Cohen wrote: what's the contents of /etc/zaptel.conf ? pbx1:~# cat /etc/zaptel.conf # # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us # fxsks=1 # channels=1 This error comes from ztcfg . Look at ztcfg.c at the tarball of zaptel. Though from the word tones I gather that this is related to the tonezone library. Unfortunately I am not a C programmer, and thus the file in question is largely shrapnel to me. However, from what I can glean, the function in question is static int rad_chanconfig(char *keyword, char *args) and has something to do with struct zt_radio_param. I am puzzled as to what is going on that it thinks a radio is involved. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --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] How can I stop lost DNS from killing Asterisk?
Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? ___ --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: Setting QOS settings in asterisk and/or CentOS?
I found this command if your Cisco switches support it: auto qos voip trust You set this on each interface. It automatically prioritizes all SIP and skinny traffic, but not iax. There is also auto qos voip cisco-phone. This one can detect a Cisco phone and prioritize it. I just have to figure out how to verify that it is actually doing anything. -- -- Steven http://www.glimasoutheast.org Rich Adamson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Nick Hoffman wrote: On Sat September 23 2006 06:14, Bob Amen [EMAIL PROTECTED] wrote: snip which sets the TOS bit on all IAX, SIP and RTP packets. Using iptables means that we can set up our rules on the router without using ACLs. Our Cisco Cookbook (http://www.oreilly.com/catalog/ciscockbk/) has a nice section on QoS (Chapter 11) and an appendix on TOS, etc. The author advises not to use ACLs when possible as they take more CPU in the router to implement and on a heavily loaded router can cause packet delays. So here's what our config looks like: snip Cheers, Bob Hi Bob. I'm new to TOS and DSCP, but after going over your and Rich Adamson's responses to Steve BerkHolz's question, I read up about them. With what you wrote above, does this mean that your Cisco router(s) deny, allow, and route traffic based on TOS/DSCP flags, and you don't bother with traditional ACL rules like below?: access-list 123 permit udp 1.2.3.4 ... ACL's in cisco hardware can be used for pattern matching in addition to the old permit, deny, etc, functions. Here's a working example from a cisco 1750 with QoS: class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any In the above, any packet matching the access-list 103 gets treated as a voice-rtp class, and in the policy map, is acted upon as priority (which means low latency queue) and can use up to 40% of the interfaces bandwidth. The bandwidth 384 statement on the interface is used by QoS to determine how much is actually going to be used for voip. interface Dialer0 bandwidth 384 ip address negotiated encapsulation ppp dialer pool 1 dialer-group 1 service-policy output voice-policy ppp pap sent-username x_dsl password 7 136775499987 That bandwidth statement should be the actual amount of bandwidth available and not the value that your dsl/broadband provider says they provide. Once the policy map is implemented, one can review the operational statistics by doing something like this: C1750#show policy-map interface dialer0 Dialer0 Service-policy output: voice-policy Class-map: voice-rtp (match-all) 1441504 packets, 191386680 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 103 Weighted Fair Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 40 (%) Bandwidth 153 (kbps) Burst 3825 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 Class-map: www-traffic (match-all) 484061 packets, 341420115 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 105 Weighted Fair Queueing Output Queue: Conversation 137 Bandwidth 30 (%) Also, by doing the following: C1750#show access-list 103 Extended IP access list 103 permit ip any any dscp cs3 permit ip any any dscp ef (1680 matches) permit ip any any tos min-delay (808709 matches) permit ip any any tos 12 (1 match) one can see which piece of an access list is being matched. One can also see that both TOS and DSCP definitions can be used within the same access list. Its kind of a handy way to ensure voip phones and asterisk are properly configure and thus properly treated from a QoS perspective. It should also be noted the above router is running v 12.2(4)T7 code. Cisco has made several changes to the syntax and parameters implemented in each version in the last few years. In the newer IOS versions (for both switches and routers), the syntax and parameters are becoming much more standardized across all product lines. The OP was specifically asking about QoS on a cisco switch, and without researching exactly what was implemented in his switch, there really isn't any way to give him a QoS template that would be accurate. For example, if I posted something that worked in the 12.4 code, its highly likely not to be acceptable syntax for 12.1 or 12.2. Whether one uses access lists to do pattern matching is mostly
Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?
Eric ManxPower Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. ___ --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] Call back
First hi
I am trying to make a call back system
I am using callme.php(click to call) to write the file at
/var/spool/asterisk/outgoing
And at the incoming context ,I match the
incoming did as follow
exten = 009613504768,4,system(elinks -dump http://127.0.0.1/click/callme.php?number=SIP/1234channel=${CHANNEL})
at callme,php I hangup the call by
system(asterisk -r -x 'soft hangup $channel' );
my troubles is I that know that its a silly
idea to make a call back system in this way,
2nd I dont hear a ringback or hangup
tone.
Please if you know a better way to that that please
dont hesitate to inform me .
Regards
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Re: [asterisk-users] Re: Setting QOS settings in asterisk and/or CentOS?
Steven wrote: I found this command if your Cisco switches support it: auto qos voip trust You set this on each interface. It automatically prioritizes all SIP and skinny traffic, but not iax. There is also auto qos voip cisco-phone. This one can detect a Cisco phone and prioritize it. I just have to figure out how to verify that it is actually doing anything. The auto qos function is a relatively new addition to the cisco routers and switches (eg, last year or so). The parameter is added to an individual interface (usually a serial interface), and it truly watches for actual traffic on that interface until you shut it down. At that point, auto qos writes the policy statements into the router config needed to support that actual traffic. To use it, you must: - enable it on an individual interface, - do not change the interface bandwidth statement while its running, - cisco express forwarding must be enabled, and, - all previously attached QoS policies must be removed from the interface being sampled. Its my understanding (although I've not actually done this) that auto qos can be used to monitor all traffic and not just voip packets. For example, some companies may wish to generate qos policies for Citrix, MS Terminal Server traffic, etc, and may not have any voip implementation at all. So, auto qos is not just for voip traffic and should be very usable with iax. Since you've specifically mentioned the auto qos voip cisco-phone statement, that statement essentially says watch for voip traffic coming from a cisco phone. Reading between the lines says: Cisco ships their voip phones with QoS already preconfigured with signaling traffic in one DSCP class and rtp traffic in another DSCP class. If your non-cisco phones aren't set up with those exact same DSCP markings, auto qos won't write the policy statements into your router's config. (E.g., cisco tends to push their proprietary voip sutff, so guess what... auto qos voip cisco-phone was oriented around those phones and not necessarily the sip versions of that same cisco phone.) The simplest command is auto qos applied to an individual interface without any other qualifying parameters. Keep in mind that auto qos is actually monitoring your traffic in real time, which assumes you've got voip phones, asterisk box, etc, already preconfigured to mark packets with TOS or DSCP bits. If that's not the case, then your voip traffic appears as default non-qos traffic and no policy will be written to the router's config. For testing purposes, auto qos can be applied to an interface then multiple voip test calls can be initiated manually. It would then write the appropriate policy statements into your config based on those voip test calls. In a large production world, one would apply auto qos to an interface and let it be for some much longer period of time (eg, hours). Then auto qos would write the config statements necessary to support the actual traffic observed over that period of time. There is no magic behind using auto qos; you can do the exact same thing manually by configuring policies in the router and doing something like show policy-map interface s1. That display will tell you how much bandwidth is consumed for each QoS class that has been configured in your policy. The problem with doing that manually is that you have to know when your peak traffic period is for voip traffic, and then run the commands during that peak period to get it right. There are technical white papers on the cisco web site (somewhere) that describes how to use the auto qos function, but keep in mind the function was only recently introduced so it is not yet implemented on every product or in every IOS image. ___ --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] How can I stop lost DNS from killing Asterisk?
Eric Bishop wrote: Hi All, When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? We just went through the same problem. You need both a caching dns server, and, define your asterisk system in /etc/hosts so he knows who he is. I've tested this several times as we use a laptop to demo asterisk and several of these demo's don't have any internet access. (And, you're right, asterisk does not process any calls.) With dns caching and the /etc/hosts definition in place, it now works everywhere. ___ --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] AGI Errors
1 - Eclipse situation - What is inside fastagi-mapping.properties ? Are you using the sample HelloAgiScript from asterisk-java ? 2 - Command line situation - what's the command line you are using ? []'s, Edmilson Santana Unitech Tecnologia de Informação (http://www.unitech.com.br/) [EMAIL PROTECTED] wrote: i have all files in the same directory: c:\agi (asterisk-java-0.2.jar, fastagi-mapping.properties, HelloAgiScript.class and HelloAgiScript.java). My slasspath is also c:\agi Did you mean this? But i get still the following errors: if i start it with eclipse: ... INFO: Received connection. 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Unable to create AGIScript instance of type HelloAgiScript 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost.ch/hello.agi' (script 'hello.agi') if i start from the console another error occurs: INFO: Received connection. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Resource bundle 'fastagi-mapping' is missing. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost/hello.agi' (scri pt 'hello.agi') What could that be? ___ --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] Is there T.38 support on asterisk 1.4 beta2 ???
Do anybody knows? Rgds, Ricardo. ___ --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] Line Pickup Problem
Pato Valarezo wrote: Lacy Moore - Aspendora wrote: Wherever you have your exten = s,1,Answer statement, replace with: exten = s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer exten = s,n,Answer then continue on. Asterisk will then wait 30 seconds before it answers the phone. You would probably want this a lower number, though. Hi, i'm using x100P clones and i have two related issues: 1. In the first system (or in both) when someone answer the call, asterisk doesn't notice the stop ringing signal and continues with the dialplan, and of course answer the call and plays the welcome message and interrupts the current call in progress. 2. One of the system wich is connected to the PSTN doesn't seems to wait the time i specify in exten = s,1,Wait(10), and answers the line in a shorter time... it seems like the time doesn't count to it. I'm testing and training with this systems until i can buy a better quality hardware i expect to not have this problems with digium or better hardware. If someone has experience in this i'll apreciate comments. Based only on the words that you've used above, it sounds like you have a problem with extensions.conf (and maybe with the 'context' associated with the x100p card. To better understand your issue, we'll need to see your extensions.conf file and zapata.conf file contents. I'd suggest not trying to copy/paste a piece of those two files but rather include the entire files. ___ --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] Play wav file during conversation
I want to be able to playback a certain soundfile for all parties in a call to hear. How would I do that? Eric ___ --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] How can I stop lost DNS from killing Asterisk?
Rich Adamson wrote: Eric ManxPower Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. A correctly set up system would already have that info in /etc/hosts, but it is a good thing to check because most systems are not correctly set up. ___ --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 - alcatel
I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira[EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA [EMAIL PROTECTED]: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch t3 to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp [EMAIL PROTECTED]: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] Re: Dual core
Here's what we set in menuconfig when building Linux kernels for multi-processor systems: Processor Type and Features --- -Symmetric multi-processing support -Timer frequency (1000 HZ) Device Drivers --- Character devices --- * Enhanced Real Time Clock Support Real Time Clock --- * RTC class MATT--- On 9/26/06, Raphaël Jacquot [EMAIL PROTECTED] wrote: Matt Florell wrote: For the Asterisk installation, no. For Linux, yes. I built a custom SMP kernel, which depending on your Linux distribution may or may not be necessary for you. what specific things have you done, that isn't in the base kernel ? ___ --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] University dumps CISCO VoIP for Asterisk
On Wednesday 20 September 2006 21:40, Douglas Garstang wrote: We stuck OpenSER in between the phones and Asterisk, and pointed our phones towards the OpenSER boxes for SIP registrations and subscriptions. When OpenSER received a REGISTER or SUBSCRIBE message, it would use the send() command to forward the messages onto each Asterisk server. By doing that, ALL of our Asterisk servers had a copy of all sip registrations and subscriptions. It seemed to work pretty well, but for unrelated reasons, we dropped that approach. Which approach do you use now? -A. ___ --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 t.38 (was RE: [asterisk-users] trixbox t38 pass through)
marek cervenka wrote: T38 passthrough doesn't seem to work in trunk at the moment. that's true http://bugs.digium.com/view.php?id=7679 http://bugs.digium.com/view.php?id=7844 t.38 in asterisk 1.4 http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I've taken the code in Openpbx somewhat farther than the code in Asterisk SVN. Openpbx is now working for a lot of T.38 passthrough scenarios, and T.38 termination is now fairly solid. T.38 gateway is also basically working, though I haven't yet handed that out to anyone else for further testing. The big thing that had to change was to reuse the RTP port for the UDPTL stream. The code I donated to * was based on the specs. We found too many things that just don't work if you simply follow the specs. To make the software tolerant of a lot of other boxes doing weird things it seems you really have to reuse the RTP port for the UDPTL stream. 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] Running Multiple Instances of Asterisk
On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote: Jay R. Ashworth wrote: voicemail.conf doesn't, as it needs to be modified by app_voicemail for password changes. An alternative is to use an external script to modify that file. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I stop lost DNS from killing Asterisk?
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? ___ --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] Running Multiple Instances of Asterisk
On Tue, Sep 26, 2006 at 06:03:46PM +0300, Tzafrir Cohen wrote: On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote: Jay R. Ashworth wrote: voicemail.conf doesn't, as it needs to be modified by app_voicemail for password changes. An alternative is to use an external script to modify that file. Careful with the quoting there, Tzafrir: (it correctly indicates) that I didn't actually say that, and I'm not the person who cares, anyway. :-) That said, thanks for at least *clipping the quotes* in much the same way that almost everyone else doesn't. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 1.4 branch and chan-capi-0.7.0
On Tue, 26 Sep 2006, Guido Hecken wrote: Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? chan-capi was not ported to Asterisk 1.4 yet. See bug http://bugs.melware.net/mantis/view.php?id=20 Armin System: Fedora Core 4 with Kernel 2.6.17-1.2142_FC4 AVM Fritz Card is present and fcpci running and up isdn4k-utils and isdn4k-utils-devel installed capi4hylafax installed make in chan_capi source said: gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:146: warning: type defaults to 'int' in declaration of 'STANDARD_LOCAL_USER' chan_capi.c:146: warning: data definition has no type or storage class chan_capi.c:147: warning: type defaults to 'int' in declaration of 'LOCAL_USER_DECL' chan_capi.c:147: warning: data definition has no type or storage class chan_capi.c: In function 'capi_new': chan_capi.c:2078: error: 'struct ast_channel' has no member named 'type' chan_capi.c: In function 'pbx_capicommand_exec': chan_capi.c:4582: warning: implicit declaration of function 'LOCAL_USER_ADD' chan_capi.c:4597: warning: implicit declaration of function 'LOCAL_USER_REMOVE' chan_capi.c: At top level: chan_capi.c:5244: error: unknown field 'send_digit' specified in initializer chan_capi.c:5244: warning: initialization from incompatible pointer type make: *** [chan_capi.o] Error 1 Thanks for any hints and ideas Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0
On Tue, 26 Sep 2006, Guido Hecken wrote: -Ursprüngliche Nachricht- Von: Armin Schindler [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 26. September 2006 13:37 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0 On Tue, 26 Sep 2006, Guido Hecken wrote: Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? chan-capi was not ported to Asterisk 1.4 yet. See bug http://bugs.melware.net/mantis/view.php?id=20 Armin Armin, thanks for the info. Are there any plans on porting it to 1.4 and if yes, is there an approximate release date? Yes, of course it is planned ;-) But currently I have much to do, so if no one else has time, it will take a little bit longer and I cannot tell a date. Armin___ --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] asterisk - alcatel
Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? 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
Re: [asterisk-users] Re: Dual core
Matt Florell wrote: For the Asterisk installation, no. For Linux, yes. I built a custom SMP kernel, which depending on your Linux distribution may or may not be necessary for you. what specific things have you done, that isn't in the base kernel ? ___ --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] Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma
Giorgio Incantalupo wrote: Hi, I have an Asterisk box with a Sangoma a102 making bridge between an old legacy PBX and a PRI line. Our customer told us he could not call some numbers so we investigated and found he was right: each time we tried to call some numbers the other party hang up. It could be be the other party PBX so we acted on zapata.conf and finally we found the right parameters configuration and we could call those numbersbut now it is not possible to receive inbound calls. Is something related to This is too less information to find your problem. What is the problem in receiving incoming calls? Activate PRI debugging set verbose 9 pri debug span 1 pri debug span 2 and watch the q931 signaling. Where is the problem with the incoming call: between PSTN and Asterisk or between Asterisk and the PBX? I guess you have problems with the Type of Number (pridialplan , prilocaldialplan ) regards klaus http://bugs.digium.com/print_bug_page.php?bug_id=7511? Why I cannot receive inbound calls?? Is there anybody experienced situation like this? TIA Giorgio Incantalupo (zapata.conf follows) ; this channels are connected to PRI context = outbound_calls group = 1 immediate = no internationalprefix = 00 language = us nationalprefix = 0 pridialplan = unknown prilocaldialplan = unknown priindication = inband resetinterval = never signalling = pri_cpe switchtype = national usecallerid = yes callerid = asreceived overlapdial=no relaxdtmf=yes usedistinctiveringdetection=yes channel = 1-15,17-31 ; this channels are connected to legacy PBX context = legacy_zap group = 2 immediate = no ;internationalprefix = 00 language = us ;nationalprefix = 0 ;pridialplan = national priindication = inband resetinterval = never signalling = pri_net switchtype = euroisdn ;usecallerid = yes callerid = asreceived relaxdtmf=yes overlapdial=yes ; YES is mandatory here usedistinctiveringdetection=yes channel = 32-46,48-62 ___ --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] ztcfg / X100P question
On Mon, Sep 25, 2006 at 09:53:24AM -0400, Michel Vaillancourt wrote: Hi, folks. I've got an X100P Wildcard here. I get an odd error when running ZTCFG on it. === pbx1:~# asterisk -V Asterisk SVN-branch-1.2-r43509 pbx1:~# lsmod Module Size Used by wcfxo 13184 0 zaptel202148 1 wcfxo crc_ccitt 2208 1 zaptel pbx1:~# dmesg | grep -i zap Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1468 Echo Canceller: KB1 pbx1:~# ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 10: Cannot get number of tones for channel 1 line 10: Cannot init tones for channel 1 2 error(s) detected what's the contents of /etc/zaptel.conf ? === I've run google on the errors, but all I turn up are Asterisk source code hunks that really don't explain to me what *triggers* that error. Could someone suggest to me what the issue could be? This error comes from ztcfg . Look at ztcfg.c at the tarball of zaptel. Though from the word tones I gather that this is related to the tonezone library. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] core dump with 1.2.7.1 and chan-capi-cm 0.6.5
Hi Armin!
My Asterisk crashes once a day. The backtrace is:
Reading symbols from /usr/lib/asterisk/modules/func_enum.so...done.
Loaded symbols for /usr/lib/asterisk/modules/func_enum.so
Reading symbols from /usr/lib/asterisk/modules/func_uri.so...done.
Loaded symbols for /usr/lib/asterisk/modules/func_uri.so
Reading symbols from /lib/libnss_dns.so.2...done.
Loaded symbols for /lib/libnss_dns.so.2
#0 capi_handle_dtmf_fax (c=0x81b5b80) at chan_capi.c:1923
1923if (p-faxhandled) {
(gdb) bt
#0 capi_handle_dtmf_fax (c=0x81b5b80) at chan_capi.c:1923
#1 0x405df41f in capi_handle_facility_indication (CMSG=0xbd9ff944,
PLCI=258, NCCI=258, i=0x818a520) at chan_capi.c:2671
#2 0x405dcb66 in capi_handle_msg (CMSG=0xbd9ff944) at chan_capi.c:3505
#3 0x405da146 in do_monitor (data=0x0) at chan_capi.c:4186
#4 0x40024e51 in pthread_start_thread () from /lib/libpthread.so.0
#5 0x401ec8aa in clone () from /lib/libc.so.6
(gdb) quit
Is this is a known problem fixed in chan_capi-cm 0.7?
thanks
klaus
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Re: [asterisk-users] Line Pickup Problem
Lacy Moore - Aspendora wrote: Wherever you have your exten = s,1,Answer statement, replace with: exten = s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer exten = s,n,Answer then continue on. Asterisk will then wait 30 seconds before it answers the phone. You would probably want this a lower number, though. Hi, i'm using x100P clones and i have two related issues: 1. In the first system (or in both) when someone answer the call, asterisk doesn't notice the stop ringing signal and continues with the dialplan, and of course answer the call and plays the welcome message and interrupts the current call in progress. 2. One of the system wich is connected to the PSTN doesn't seems to wait the time i specify in exten = s,1,Wait(10), and answers the line in a shorter time... it seems like the time doesn't count to it. I'm testing and training with this systems until i can buy a better quality hardware i expect to not have this problems with digium or better hardware. If someone has experience in this i'll apreciate comments. -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com Res non verba. (Las vacas no hablan). -- Cantervill. (1953) Poeta, cantautor, internauta y webmaster argentino. ___ --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] ztcfg / X100P question
On Tue, Sep 26, 2006 at 08:45:49AM -0400, Michel Vaillancourt wrote: Tzafrir Cohen wrote: what's the contents of /etc/zaptel.conf ? pbx1:~# cat /etc/zaptel.conf # # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us # fxsks=1 # channels=1 This line is unnecessary. Just remove it. This error comes from ztcfg . Look at ztcfg.c at the tarball of zaptel. Though from the word tones I gather that this is related to the tonezone library. Unfortunately I am not a C programmer, and thus the file in question is largely shrapnel to me. However, from what I can glean, the function in question is static int rad_chanconfig(char *keyword, char *args) and has something to do with struct zt_radio_param. I am puzzled as to what is going on that it thinks a radio is involved. The keyword channels you used is for some radio-related stuff. You configured zaptel.conf and zapata.conf... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: does /var/run/asterisk.ctl exist? -- butAsterisk *is* running.
I had a problem on one box where /var/run/asterisk/ did exist and had the correct non-root permissions. There was a typo in /etc/asterisk/asterisk.conf. was: astrundir = /var/run changed to: astrundir = /var/run/asterisk I do not remember which version of asterisk this was or if it was broken when adding freepbx. (or maybe when I tested [EMAIL PROTECTED]) I hope this helps. -- -- Steven http://www.glimasoutheast.org "Lacy Moore - Aspendora" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... Be sure that it is looking in the right place. If it is running as non root, then the ctl file would be in a different directory. It looks as though Trixbox does run as non-root. The ctl is actually/var/run/asterisk/asterisk.ctl. Did you install from scratch, or was a previous version of Asterisk on the box? On 9/25/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Sorry if I'm stating the obvious, but I'm not sure if Trixbox runsasterisk as root or not.I have to "sudo asterisk -r" on mine, but I'm not running Trixbox, I'm running Asterisk 1.2.MojKen D'Ambrosio wrote: I've set up a bunch of plain-jane Asterisk systems, but had heard good things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox. So I installed it, and fired it up, and it works fine. Until I try to do an "asterisk -r".I get the "does /var/run/asterisk.ctl exist?" question, which had always previously meant (to me) that Asterisk wasn't running.But it is!And there's now asterisk.ctl file in the entire /var hierarchy.Anyone have any ideas as to why that might be MIA?It's insanely annoying, not being able to fire up the console. Thanks, -Ken ___ --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 !DSPAM:500,451866a1139541174510073! --Mojo [EMAIL PROTECTED]Office Manager, Horan Company, LLC(907) 747- x112___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] OT: Opinions on Aastra 480i CT?
Hang onI have a 480i on my desk. The deskset definitely has a hold key. The programmable keys make a VM key really easy too. The cordless handset is limited by the number of buttons, but there are keystrokes for hold and a number of other functions. I wouldn't say that the cordless could be your main phone, but it certainly suuports the deskset well. Michael On Mon, 25 Sep 2006 09:31:42 -0600, Colin Anderson wrote: It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! aw crap, that's a biggie but I think I can work around it, teach the user to dial *98 for voicemail, *700 for park and hash to transfer, currently the users dial feature-9-8-1 for voicemail right now so they are used to doing things the hard way. But a dedicated hold and transfer button would've been nice. The users' big requirement is inbound /outbound / missed call logging, how is that? ___ --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] Set hint status from dialplan?
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
http://bugs.digium.com/view.php?id=5779
On 9/26/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
Is it possible to manually set the hint status of a virtual extension via
the dialplan?
I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn on a
light for night mode.
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
for more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activations
exten = 799,hint,DS/mmgc
exten =
799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)
exten = 799,n,Playback(beep)
exten = 799,n,Hangup
[macro-open-close]
exten = s,1,DBGet(nightmode=nightmode/${ARG1})
exten = s,n,NoOp(ARG1 ${ARG1} ARG2 ${ARG2} ARG3 ${ARG3})
exten = s,n,Set(OpenFile=${ARG2})
exten = s,n,Set(CloseFile=${ARG3})
exten = s,n,NoOp(Close file ${CloseFile}. Open file ${OpenFile}.)
exten = s,n,GotoIf(${nightmode}=1?s,Open:s,Close)
exten = s,n(Open),DBPut(nightmode/${ARG1}=0)
exten = s,n,Devstate(${ARG1},0)
exten = s,n,Playback(${OpenFile})
exten = s,n,Goto(Return)
exten = s,n(Close),DBPut(nightmode/${ARG1}=1)
exten = s,n,Devstate(${ARG1},2)
exten = s,n,Playback(${CloseFile})
exten = s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it turns
night mode on and announces that, another press and it turns night mode off
and announces that..
--
Lacy Moore
I'm the guy that doesn't give a damn about anyone's problems but my own...
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Re: [asterisk-users] Re: asterisk to cell phone network
On 10:25, Tue 26 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1/2/4 simslot pci card: http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. Where can I see the prices of this cards? My supplier has them listed as: UnoGSM: 900 euro DuoGSM: 1200 euro QuadGSM: 1600 euro -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] X100P Clone card in JAPAN
Did anyone success to install X100P Clone card on Asterisk to work with Japan stanadards for Analog line over ISDN TA. I can't make call when the call is ringing is OK and in the moment when the call is pick up the line is droped. I have hang ups all the time. I can't make call. Did anyone else have this problem. Is it depends on the Japanes provider or ? Help Please. ___ --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: asterisk to cell phone network
Ciao Michiel, http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. Where can I see the prices of this cards? My supplier has them listed as: UnoGSM: 900 euro DuoGSM: 1200 euro QuadGSM: 1600 euro Well, how does Asterisk interact with those devices? Is there a chan_gsm_pci? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Asterisk behind Sonicwall firewall
Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Thanks all ___ --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 behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? SIP is still on 5060, but the AUDIO (which is RTP) is on a dynamically negotiated port. Now you understand why many people in the VoIP business would love to meet the people that designed SIP in a dark alley. Read the mailing list archives and the Wiki for information working around these issues. ___ --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 behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Thanks all ___ --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 http://www.lassologic.com/support/pdfs/Configuring_Voip_For_SonicOS_Enhanced.pdf#search=%22sonicos%20voip%22 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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: Set hint status from dialplan?
;exten = 799,hint,DS/mmgc
Lacy, What is the DS/mmgc?
-- -- Steven
http://www.glimasoutheast.org
"Lacy Moore - Aspendora" [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]...
Is
it possible to manually set the hint status of a virtual extension via the
dialplan?I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn
on a light for night mode.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstatefor
more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activationsexten = 799,hint,DS/mmgcexten =
799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)exten
= 799,n,Playback(beep)exten = 799,n,Hangup
[macro-open-close]exten =
s,1,DBGet(nightmode=nightmode/${ARG1})exten = s,n,NoOp(ARG1 ${ARG1}
ARG2 ${ARG2} ARG3 ${ARG3})exten = s,n,Set(OpenFile=${ARG2})exten
= s,n,Set(CloseFile=${ARG3}) exten = s,n,NoOp(Close file
${CloseFile}. Open file ${OpenFile}.)exten =
s,n,GotoIf(${nightmode}=1?s,Open:s,Close)exten =
s,n(Open),DBPut(nightmode/${ARG1}=0)exten = s,n,Devstate(${ARG1},0)
exten = s,n,Playback(${OpenFile})exten =
s,n,Goto(Return)exten = s,n(Close),DBPut(nightmode/${ARG1}=1)exten
= s,n,Devstate(${ARG1},2)exten =
s,n,Playback(${CloseFile})exten = s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it
turns night mode on and announces that, another press and it turns night mode
off and announces that..
-- Lacy MooreI'm the guy that doesn't give a
damn about anyone's problems but my own...
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Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Which Sonicwall model? Some (like the TZ170) have special VOIP settings, like Enable consistent NAT and Enable SIP Transformations. Check those; they work well with SIP. If you don't have one of these newer models, please see http://www.voip-info.org/wiki-IAX, in the NAT Issues section. It deals with IAX2, but the issues are same for SIP UDP. The Sonicwall UDP-connection-memory timeout may be VERY short - 30 seconds by default on some! It is adjustable in some firmware versions. I use the TZ170, but with IAX2 rather than SIP. - Mike ___ --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: Set hint status from dialplan?
That is the metermaid patch. It has been included into 1.4 as far as I know.
I am hoping to use that for parking slot BLFs on the phones.
My extension for day/night mode is not a real channel, so I am hoping to set
the hint value manually.
--
--
Steven
http://www.glimasoutheast.org
C F [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
http://bugs.digium.com/view.php?id=5779
On 9/26/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
Is it possible to manually set the hint status of a virtual extension via
the dialplan?
I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn on a
light for night mode.
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
for more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activations
exten = 799,hint,DS/mmgc
exten =
799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)
exten = 799,n,Playback(beep)
exten = 799,n,Hangup
[macro-open-close]
exten = s,1,DBGet(nightmode=nightmode/${ARG1})
exten = s,n,NoOp(ARG1 ${ARG1} ARG2 ${ARG2} ARG3 ${ARG3})
exten = s,n,Set(OpenFile=${ARG2})
exten = s,n,Set(CloseFile=${ARG3})
exten = s,n,NoOp(Close file ${CloseFile}. Open file ${OpenFile}.)
exten = s,n,GotoIf(${nightmode}=1?s,Open:s,Close)
exten = s,n(Open),DBPut(nightmode/${ARG1}=0)
exten = s,n,Devstate(${ARG1},0)
exten = s,n,Playback(${OpenFile})
exten = s,n,Goto(Return)
exten = s,n(Close),DBPut(nightmode/${ARG1}=1)
exten = s,n,Devstate(${ARG1},2)
exten = s,n,Playback(${CloseFile})
exten = s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it turns
night mode on and announces that, another press and it turns night mode off
and announces that..
--
Lacy Moore
I'm the guy that doesn't give a damn about anyone's problems but my own...
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Re: [asterisk-users] Re: Set hint status from dialplan?
On Tuesday 26 September 2006 13:02, Steven wrote: That is the metermaid patch. It has been included into 1.4 as far as I know. I do not see DevState in my show application output, so I would say no, it's not in 1.4. -A. ___ --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] PRI Outbound CallerID Question
Hi all, I've searched around and haven't found much of an answer to my issue. Any advice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to another PSTN user via the PRI, sending the original callers id with it. I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcall to the pstn numbers we have defined and the incoming call shows up with the original callers CID, we answer and have options to accept or reject the call. So I know the 800 provider is staying in the middle of the call and not just performing a redirect to us. I've tried the various CID settings in Asterisk, but am not able to use anything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue. Does anyone know if it's possible with a PRI or do you have to have some other type of PSTN connection such as SS7? Thanks!! --Shawn ___ --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: Set hint status from dialplan?
;exten = 799,hint,DS/mmgc Lacy, What is the DS/mmgc? The DS is what the DevState patch adds. I actually got to this point by following this thread: http://forums.digium.com/viewtopic.php?t=891highlight=shared+line After I implemented these changes, I had the DevState on the system. The mmgc is just an arbitrary name you can use. We have several companies sharing the same phone system, and that is one of the companies. The Polycom monitors the hint status of extension 799, which is the Device State of mmgc. ___ --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: Set hint status from dialplan?
Steven, If youare trying to do this on a stock Asterisk system (and I can certainly understand why you would want to), then what I have implemented will definitely not work. I couldn't find anyway to do this on a stock system. Upgrades are going to be a nightmare with all the patches that have been applied to my system. This was something that was absolutely needed, and the patch that started me down that road was something that was absolutely needed. I had to have a completely dummy proof way for someone to park a call and pick it up at another extension. When I say dummy proof, I mean it, too :-) That patches mentioned in the forum link do that very well, and as an added bonus, took care of my night mode indicator. ___ --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] Included context
Hello,
For example I have this dialplan:
[context1]
exten = s,1,Noop
exten = s,n,Dial(...)
exten = s,n,Playback(${CONTEXT})
exten = s,n,Hangup
[context2]
include = context1
[context3]
include = context1
Then I make dial-out call files with context2, context3, etc. What is
the value of ${CONTEXT} in that case? Still context1 because it's
physically there or context2, context3 because I am included from
there (so in a way logically is there). I didn't find any exact answer
that's why I'm asking here.
bye,
Zsolt
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Re: [asterisk-users] PRI Outbound CallerID Question
Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs). On 9/26/06, Shawn Kelley [EMAIL PROTECTED] wrote: Hi all,I've searched around and haven't found much of an answer to my issue. Anyadvice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to anotherPSTN user via the PRI, sending the original callers id with it.I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcallto the pstn numbers we have defined and the incoming call shows up with theoriginal callers CID, we answer and have options to accept or reject the call.So I know the 800 provider is staying in the middle of the call and not justperforming a redirect to us.I've tried the various CID settings in Asterisk, but am not able to useanything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue.Does anyone know if it's possible with a PRI or do you have to have someother type of PSTN connection such as SS7?Thanks!!--Shawn ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --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 Gateway
I am thinking of using a mini atx 1u server with a digium zaptel (wcte11xp) installed to act as a SIP gateway. This way any of my asterisk servers can forward calls to any gateway (seperated by about 3miles of fiber). Has anyone else tried this? I would just load a basic asteisk config and zaptel with something like CentOS 4.4 ServerCD. Here is the hardware I am thinking of. http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html It seems like this would be alot cheaper than getting a pre-built sip gateway from VOX. Any input is greatly appreciated. Forrest ___ --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] PRI Outbound CallerID Question
Besides for what Lacy answered, have you tried NOT playing with setting CID? Just do a blind xfer, or just use dial whatever on the DID itself. If that doesn't work then like Lacy said your provider might be blocking it. On 9/26/06, Shawn Kelley [EMAIL PROTECTED] wrote: Hi all, I've searched around and haven't found much of an answer to my issue. Any advice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to another PSTN user via the PRI, sending the original callers id with it. I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcall to the pstn numbers we have defined and the incoming call shows up with the original callers CID, we answer and have options to accept or reject the call. So I know the 800 provider is staying in the middle of the call and not just performing a redirect to us. I've tried the various CID settings in Asterisk, but am not able to use anything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue. Does anyone know if it's possible with a PRI or do you have to have some other type of PSTN connection such as SS7? Thanks!! --Shawn ___ --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] Re: Set hint status from dialplan?
On Tuesday 26 September 2006 13:25, Lacy Moore - Aspendora wrote: http://forums.digium.com/viewtopic.php?t=891highlight=shared+line Direct link for those of us who can't stand forums: http://bugs.digium.com/view.php?id=5779 -A. ___ --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] PRI Outbound CallerID Question
There seems to be three tiers in my experience: 1. Only your DID's 2. Arbitrary, but the pilot number of the PRIwill appear if you suppress your Caller ID 3. Completely arbitrary, including null --this is the fa shizzle So you want 2) or 3) but definitely it is a telco thing. You need to sweetly social engineer someone in the call centre at your telco. -Original Message-From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED]Sent: Tuesday, September 26, 2006 11:50 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] PRI Outbound CallerID Question Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs). On 9/26/06, Shawn Kelley [EMAIL PROTECTED] wrote: Hi all,I've searched around and haven't found much of an answer to my issue. Anyadvice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to anotherPSTN user via the PRI, sending the original callers id with it.I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcallto the pstn numbers we have defined and the incoming call shows up with theoriginal callers CID, we answer and have options to accept or reject the call.So I know the 800 provider is staying in the middle of the call and not justperforming a redirect to us.I've tried the various CID settings in Asterisk, but am not able to useanything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue.Does anyone know if it's possible with a PRI or do you have to have someother type of PSTN connection such as SS7?Thanks!!--Shawn___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --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] Included context
How about trying Noop(CONTEXT IS: ${CONTEXT}) before you post it? but
now that you have posted just try it and report back.
On 9/26/06, Artifex Maximus [EMAIL PROTECTED] wrote:
Hello,
For example I have this dialplan:
[context1]
exten = s,1,Noop
exten = s,n,Dial(...)
exten = s,n,Playback(${CONTEXT})
exten = s,n,Hangup
[context2]
include = context1
[context3]
include = context1
Then I make dial-out call files with context2, context3, etc. What is
the value of ${CONTEXT} in that case? Still context1 because it's
physically there or context2, context3 because I am included from
there (so in a way logically is there). I didn't find any exact answer
that's why I'm asking here.
bye,
Zsolt
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Re: [asterisk-users] Re: Set hint status from dialplan?
Andrew what does show channeltypes give you? On 9/26/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 26 September 2006 13:02, Steven wrote: That is the metermaid patch. It has been included into 1.4 as far as I know. I do not see DevState in my show application output, so I would say no, it's not in 1.4. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Thanks All I have those settings already enabled It is rejecting the SIP INVITE packet not even getting to Voice at all The VoIP provider shows a registered with a good Qualify time 55 ms but not calls come in due to the Invite packet being rejected Why and why would it suddenly do this nothing was changed it blink I'm not going to work now Thanks all Barry Dr. Michael J. Chudobiak wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Which Sonicwall model? Some (like the TZ170) have special VOIP settings, like Enable consistent NAT and Enable SIP Transformations. Check those; they work well with SIP. If you don't have one of these newer models, please see http://www.voip-info.org/wiki-IAX, in the NAT Issues section. It deals with IAX2, but the issues are same for SIP UDP. The Sonicwall UDP-connection-memory timeout may be VERY short - 30 seconds by default on some! It is adjustable in some firmware versions. I use the TZ170, but with IAX2 rather than SIP. - Mike ___ --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] H323 IP phones
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appearance, end-user feedback, any infowill be appreciated.thnx!Alyed ___ --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 with cisco 7935
I spent quite a bit of time debugging the 7935/7936, and it is an issue inside the firmware that Cisco knows how to work around in CallManager. There are better conference phone options available, and development on chan_sccp is basically dead at this point anyway, so I dont see this one ever being fixed. I would recommend a Polycom IP4000, its the exact same phone body but is much cheaper MSRP, and its SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, September 26, 2006 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with cisco 7935 Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it?on the The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP firmware that was releasedthis month. I don't know if it works any better. -- Lacy Moore I'm the guy that doesn't give a damn about anyone's problems but my own... ___ --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: Set hint status from dialplan?
On Tuesday 26 September 2006 13:57, C F wrote: Andrew what does show channeltypes give you? *CLI show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no -- 6 channel drivers registered. *CLI show version Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running Linux on 2006-09-12 03:02:05 UTC Curious... I see Local/ has a devicestate, and I've never heard of a Feature/ channel type before... :-) So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot state, but nothing for arbitrary channels such as what Lacy is showing. Is that correct? -A. ___ --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] Rewriting CID number w/o changing CDR src field
Hi all. As a convieneince to my users, I'm trying to strip off the leading 1 and areacode from incoming calls. However, when I do, the src field in the CDR is also stripped. I'd like the CDR to reflect the connonical form of the incoming number. Any way do to this? TIA, Mike Diehl. ___ --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 with cisco 7935
I would also recommend either the Polycom IP4000, or the Clearone MAXIP, both of which are SIP native. If cost is an issue, you can also take an inexpensive Polycom analog conference phone, such as the Voicestation 100, and SIP enable it using a Linksys SPA-1001 analog adapter. For about half the price of a native SIP conference phone you have a working solution. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Amos Sent: Tuesday, September 26, 2006 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk with cisco 7935 I spent quite a bit of time debugging the 7935/7936, and it is an issue inside the firmware that Cisco knows how to work around in CallManager. There are better conference phone options available, and development on chan_sccp is basically dead at this point anyway, so I dont see this one ever being fixed. I would recommend a Polycom IP4000, its the exact same phone body but is much cheaper MSRP, and its SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, September 26, 2006 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with cisco 7935 Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it?on the The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP firmware that was releasedthis month. I don't know if it works any better. -- Lacy Moore I'm the guy that doesn't give a damn about anyone's problems but my own... ___ --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] voicemailmain menu
Hi, Is there way a way to restrict access to certain menus, such as the following: 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Record your temporary message (new in Asterisk v1.2) Thanks in advance, Jack ___ --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: Set hint status from dialplan?
Here is an output from a 1.4.0-Beta2 voipgw*CLI show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Agent Call Agent Proxy Channel yes yes no Console OSS Console Channel Driver no yes no Zap Zapata Telephony Driver w/PRIno yes no Skinny Skinny Client Control Protocol (Skinny) no yes no Phone Standard Linux Telephony API Driver no yes no Feature Feature Proxy Channel Driver no yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MGCPMedia Gateway Control Protocol (MGCP)yes yes no -- 10 channel drivers registered. voipgw*CLI show version Asterisk 1.4.0-beta2 built by root @ voipgw on a i686 running Linux on 2006-09-25 00:49:44 UTC voipgw*CLI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, September 26, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Set hint status from dialplan? On Tuesday 26 September 2006 13:57, C F wrote: Andrew what does show channeltypes give you? *CLI show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no -- 6 channel drivers registered. *CLI show version Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running Linux on 2006-09-12 03:02:05 UTC Curious... I see Local/ has a devicestate, and I've never heard of a Feature/ channel type before... :-) So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot state, but nothing for arbitrary channels such as what Lacy is showing. Is that correct? -A. ___ --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] TE406P not working on Intel D101Ggc motherboard.
I recently moved a TE406P card from an Intel D865GBF motherboard (where it worked fine), to an Intel D101Ggc card, and now I can't get the spans to got up correctly. All I get is an endless burst of: == Primary D-Channel on span 4 up == Primary D-Channel on span 2 up !! Got a UA, but i'm in state 1 The other 2 spans doesn't even make an attempt to start. If I set up one of the semi-awake spans in intense debug mode, all that I get is: Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (0) [ 02 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter and so on... The spans are connected to Pika Primenet E1 card, which is set as Network side in the four spans. This card is installed in a different machine. My configuration for the Digium card is: * zaptel.conf: span=1,1,0,ccs,hdb3,yellow bchan=1-15 dchan=16 bchan=17-31 span=2,2,0,ccs,hdb3,yellow bchan=32-46 dchan=47 bchan=48-62 span=3,3,0,ccs,hdb3,yellow bchan=63-77 dchan=78 bchan=79-93 span=4,4,0,ccs,hdb3,yellow bchan=94-108 dchan=109 bchan=110-124 loadzone=cl defaultzone=cl * zapata.conf: [channels] language=es switchtype=euroisdn pridialplan=national prilocaldialplan=national priindication=inband usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-3.0 txgain=-15.0 canpark=yes resetinterval=never signalling=pri_cpe group=1 context=zap_pstn channel = 1-15 channel = 17-46 channel = 48-77 channel = 79-108 channel = 110-124 * Zaptel version 1.2.9.1, Asterisk 1.2.12.1, RHEL4 2.6.9-42.0.2.ELsmp, /proc/interrupts says: CPU0 CPU1 0: 47653469 47665247IO-APIC-edge timer 1: 8 0IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 67 0IO-APIC-edge i8042 14: 446945 446248IO-APIC-edge ide0 169: 0 0 IO-APIC-level libata 177: 30424 29392 IO-APIC-level libata 185: 0 0 IO-APIC-level ehci_hcd, ohci_hcd, ohci_hcd 193:222 0 IO-APIC-level HDA Intel 201: 47669502 47599256 IO-APIC-level wct4xxp 209: 129534 0 IO-APIC-level eth0 NMI: 0 0 LOC: 95329567 95329645 ERR: 0 MIS: 0 IRQ balance is running. Thanks a lot for your answers. -- Atly. Alvaro Palma. ___ --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 behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording if you mean a voip connection suddenly changed from dup/5060, or, did you change the asterisk system to use some other udp port. The sonicwall does have an option to support sip (udp/5060), but I've not had to use it on anything that we've worked with. ___ --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] Problem with zaptel 1.4b2 and X101P Wildcard
On 9/26/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Next thing to do, I guess, is to run:strace ztcfgto see which device exactly is accessed . Though /dev/zap/ctl is the
usual suspect.
[EMAIL PROTECTED] zaptel-1.4.0-beta1]# strace ztcfgexecve(/sbin/ztcfg, [ztcfg], [/* 25 vars */]) = 0uname({sys=Linux, node=mythtv, ...}) = 0brk(0) = 0x8779000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or directory)open(/etc/ld.so.cache, O_RDONLY) = 3fstat64(3, {st_mode=S_IFREG|0644, st_size=87164, ...}) = 0old_mmap(NULL, 87164, PROT_READ, MAP_PRIVATE, 3, 0) = 0xf6fea000
close(3) = 0open(/lib/tls/libm.so.6, O_RDONLY) = 3read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\0\363C..., 512) = 512fstat64(3, {st_mode=S_IFREG|0755, st_size=215248, ...}) = 0
old_mmap(0x43c000, 139424, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x43c000old_mmap(0x45d000, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x45d000close(3) = 0
open(/lib/tls/libc.so.6, O_RDONLY) = 3read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0 \0372\000..., 512) = 512fstat64(3, {st_mode=S_IFREG|0755, st_size=1512400, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fe9000old_mmap(0x30d000, 1207532, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x30d000old_mmap(0x42e000, 16384, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x12) = 0x42e000
old_mmap(0x432000, 7404, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x432000close(3) = 0old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fe8000
mprotect(0x42e000, 8192, PROT_READ) = 0mprotect(0x45d000, 4096, PROT_READ) = 0mprotect(0x309000, 4096, PROT_READ) = 0set_thread_area({entry_number:-1 - 6, base_addr:0xf6fe86c0, limit:1048575, seg_32bit:1, contents:0, read_exec_only:0, limit_in_pages:1, seg_not_present:0, useable:1}) = 0
munmap(0xf6fea000, 87164) = 0open(/dev/zap/ctl, O_RDWR) = 3brk(0) = 0x8779000brk(0x879a000) = 0x879a000open(/etc/zaptel.conf, O_RDONLY) = 4
fstat64(4, {st_mode=S_IFREG|0644, st_size=9532, ...}) = 0mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fff000read(4, #\n# Zaptel Configuration File\n#\n..., 4096) = 4096
read(4, he list remains idle\n# \clear\ ..., 4096) = 4096read(4, le is a single tone DCS transmit..., 4096) = 1340read(4, , 4096) = 0close(4) = 0
munmap(0xf6fff000, 4096) = 0ioctl(3, 0x80844a05, 0xfef3aad0) = -1 EINVAL (Invalid argument)ioctl(3, 0x404c4a13, 0x807a7cc) = -1 ENOTTY (Inappropriate ioctl for device)write(2, ZT_CHANCONFIG failed on channel ..., 71ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
) = 71close(3) = 0exit_group(1) = ?
/Morten
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[asterisk-users] Re: Running Multiple Instances of Asterisk
DG == Douglas Garstang [EMAIL PROTECTED] writes: DG I'd like to know if anyone has sucessfully managed to run multiple DG instances of Asterisk on the same system. - Did you run each DG instance as a separate user? - Did you have any install or config DG problems? - It looks like the G729 codec registration utility DG doesn't work when files aren't installed in standard places. Did DG you have this problem? - How many instances could be run on a DG single Asterisk box? We run multiple asterisks with vserver. It has a slight disk space and memory penalty, but nothing compared to proper paravirtualisation or true virtualisation. It works very well. They are SIP only though -- it would be a bit more difficult if there was hardware involved. Most people on this list think one asterisk instance will cut it for tens or even hundreds of business customers. Best of luck to them, of course. I guess the first thing they do is replace callgroups and pickupgroups with something else. /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 1.4 mohsuggest
Ok, so does anyone know who the contributor of the new moh code is into Asterisk 1.4? I'll email them directly. Doug. -Original Message- From: Douglas Garstang Sent: Tuesday, September 26, 2006 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 mohsuggest I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of documentation isn't helping much. I have this in sip.conf: [3254101] type=friend ... mohsuggest=class1 [3254102] type=friend ... mohsuggest=class2 A call is bridged between the two extensions. When 3254102 puts 3254101 on hold, 3254101 hears moh class 'class2' which is correct. However, when 3254101 puts 3254102 on hold, the 3254102 hears the default music class. Why? Doug. ___ --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] Re: asterisk to cell phone network
On 18:32, Tue 26 Sep 06, Andrea Spadaccini wrote: Well, how does Asterisk interact with those devices? Is there a chan_gsm_pci? It's using chan_zap. junghanns.net created an extra zap driver for it, same as with their quad/octobri and zap_hfc stuff. So asterisk will see it as Zap/some number -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 12:49:34PM -0500, Lacy Moore - Aspendora wrote: Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs). Specifically, carriers who permit you to connect using a technology which allows you to send originating CNID (which is basically limited to ISDN at the moment, I believe) *are supposed to* filter the CNID you present before passing it along (I believe this to be in Part 68, but can't cite it), but not all of them do. In the past, 5ESS's automatically filtered, and DMS-100's automatically didn't, though either could -- I think -- be datafilled on a trunkgroup basis to work the other way. In the OP's situation, if his carrier doesn't already forward the CNID he supplies them, then he'll likely have to sign something with the to get authorization to do it. Or, like someone said, pretext it. Oh, my; that's a bad word this year. :-) And it's not real rugged either. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 behind Sonicwall firewall
Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no real connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry Rich Adamson wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording if you mean a voip connection suddenly changed from dup/5060, or, did you change the asterisk system to use some other udp port. The sonicwall does have an option to support sip (udp/5060), but I've not had to use it on anything that we've worked with. ___ --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] TE406P not working on Intel D101Ggc motherboard.
On Tue, Sep 26, 2006 at 03:03:23PM -0400, ?lvaro Palma wrote: I recently moved a TE406P card from an Intel D865GBF motherboard (where it worked fine), to an Intel D101Ggc card, and now I can't get the spans to got up correctly. All I get is an endless burst of: As much of a pain as it is, I always as, in such circumstances: did you put it back on the original working mobo? Does it still work? They don't call it provocative maintenance for nothing. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 behind Sonicwall firewall
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no real connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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 - alcatel
Hello, We have test this configuration but we think it's a problem with the Alcatel.how are you doing to make the trunk between alcatel and Asterisk?We use a card PRA recommended by an Alcatel's technician and you? ThanksNicolasOn 9/26/06, Frederico Madeira [EMAIL PROTECTED] wrote: I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira [EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA [EMAIL PROTECTED]: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch t3 to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp [EMAIL PROTECTED]: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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
