RE: [Asterisk-Users] External relay triggered by Asterisk extension -question
Done something similar in a different application, but * should handle it -- In my case, I took a crystalfontz LCD, type 633, and used two of the four fan-outputs to drive two 12V relays. As a nice extra, you get temperature capabilities thrown in, so you can monitor your set-up. The LCD runs on serial, of course. As an alternative, you can use any of the many available relay boards -- $50 gets you this: http://www.phanderson.com/iom141.html > -Original Message- > From: James Bean [mailto:[EMAIL PROTECTED] > Sent: Saturday, February 19, 2005 11:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] External relay triggered by > Asterisk extension -question > > > > Has anyone every setup an external open/close relay, off say > a serial interface, and have an extension trigger the relay? > > Why I ask is I have a student accomodation where I am > installing an asterisk box to supply phone services to the > tenants, there is already an intercom system in the main > hallways that triggers the downstairs door and gate using a > standard relay open/close trip, so I was hoping to get the > linux box with asterisk to trip the same type of relay. > > Is there any door phones that are speaker driven only and sip > based that anyone knows about as well? > > James > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simulated dialtone like in other PBX
Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soundcard problems?
Try using ALSA with asterisk. Edit your /etc/asterisk/modules.conf And comment and uncomment lines to leave as: ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; ;noload => chan_alsa.so noload => chan_oss.so i hope this help Ariel Pablo Klein -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 4:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Soundcard problems? Has anybody had any problems with their soundcards like this: Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on sound device: Resource temporarily unavailable This shows on the console and I have no clue what it is.. voice prompts sound good Any clues? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trouble with SIP softphone calling IAX2 softphone
hi all, I have an SIP softphone (kphone on FreeBSD) and an IAX2 client (FireFly on Windows) trying to call each other. When the SIP client calls IAX client the call gets connected but the SIP client cannot hear any voice. the IAX client can hear SIP clients voice very clearly. When the IAX client tries to call the SIP client asterisk says "Unable to create a channel of type (SIP/1009)". Both of them can access Zaptel Interfaces and make calls to the PSTN. Both can call clients of their own protocol and converse. I am using ulaw and Asterisk to SIP is not behind a NAT. IAX2 is a client on an external network that successfully connects to asterisk and makes calls to the PSTN lines. I would appreciate any help what so ever. If you need me to post my confs please tell me so and I will upload them. regards Kavit ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
Yes, this is basically the default. Jon. On Sunday 20 February 2005 02:20 am, Anton Krall wrote: > Guys.. > > Im new to asterisk but is it possible to simulate a dialtone for example, > in other PBX when you pick up the phone you can hear a certain dialup, > which is the PBX dialtone, and when you hit 9, you can hear the PSTN > dialtone, is this possible? > > __ > Anton Krall > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
On Sun, 20 Feb 2005, Anton Krall wrote: > Im new to asterisk but is it possible to simulate a dialtone for example, in > other PBX when you pick up the phone you can hear a certain dialup, which is > the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is > this possible? I'm not sure I understand your question. Do you want to be able to hit 9 and get a an outside line with dialtone? Just add an extension to do that. For isdn you need to enable overlap dialing. Or do you want Asterisk to provide a dialtone after the user have hit 9 as the first digit of a number? User the ignorepat option in the dialplan. Or do you want Asterisk to provide a _different_ dialtone after the user have hit 9 as the first digit of a number? This may be possible, but I think some hack may be needed. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bridging iaxtel calls to PSTN
Hello, I just started using asterisk, and have a question. I have setup two asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1 FSX modules) and is connected to the PSTN. B has same, but is NOT connected to PSTN. I want to configure B to call A via iaxtel, and connect to the PSTN using A's line. How can I configure iaxtel dial plan for B in extensions.conf? I want to be able to make a call to local US number (where A is located) from B, using iaxtel. Can anyone please help me? All I have seen so far is just making calls from A to B and vice versa using the iaxtel 1700 number, but I haven't seen any examples of how to bridge the iaxtel calls to PSTN. Help please. chuks. NB: I don't mean toll free number, I mean just local dialing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soundcard problems?
Thx Ariel, Ill try that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Pablo Klein Sent: Domingo, 20 de Febrero de 2005 02:38 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Soundcard problems? Try using ALSA with asterisk. Edit your /etc/asterisk/modules.conf And comment and uncomment lines to leave as: ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; ;noload => chan_alsa.so noload => chan_oss.so i hope this help Ariel Pablo Klein -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 4:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Soundcard problems? Has anybody had any problems with their soundcards like this: Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on sound device: Resource temporarily unavailable This shows on the console and I have no clue what it is.. voice prompts sound good Any clues? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] making ASTCC web page secure ???
How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password, however since it is a cgi script that does not seem to be posible. Any ideas, any suggestions ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Domingo, 20 de Febrero de 2005 03:43 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX On Sun, 20 Feb 2005, Anton Krall wrote: > Im new to asterisk but is it possible to simulate a dialtone for > example, in other PBX when you pick up the phone you can hear a > certain dialup, which is the PBX dialtone, and when you hit 9, you can > hear the PSTN dialtone, is this possible? I'm not sure I understand your question. Do you want to be able to hit 9 and get a an outside line with dialtone? Just add an extension to do that. For isdn you need to enable overlap dialing. Or do you want Asterisk to provide a dialtone after the user have hit 9 as the first digit of a number? User the ignorepat option in the dialplan. Or do you want Asterisk to provide a _different_ dialtone after the user have hit 9 as the first digit of a number? This may be possible, but I think some hack may be needed. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. For SIP, you have to understand that in most situations, Asterisk will not get a dialstring until the phone decides that you are done dialling. So if you want a new dialtone after the 9, you got to configure the phone that way - if possible. Or dial 9, direct 9 to the disa() app in your dial plan and provide a new dialtone from the PBX... For local ZAP channels, the story is different, because there Asterisk will provide the dialtone. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
Anton Krall wrote: > I think it would be your last suggestion.. When I pickup the phone I hear a > tone, the sip phone box tone... Then I hit 9, no tones :) and enter the > whole phone number and it starts to ring on the other side.. So no outside > dialtone get heard ever.. I was wondering if it could be possible to make it > so that after hitting 9.. The tone would change to something else letting > the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "In the long run the pessimist may be proved right, but the optimist has a better time on the trip." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External relay triggered by Asterisk extension-question
Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] External relay triggered by > Asterisk extension-question > > Done something similar in a different application, but * > should handle it -- > > In my case, I took a crystalfontz LCD, type 633, and used two > of the four fan-outputs to drive two 12V relays. As a nice > extra, you get temperature capabilities thrown in, so you can > monitor your set-up. The LCD runs on serial, of course. > > As an alternative, you can use any of the many available > relay boards -- $50 gets you this: > http://www.phanderson.com/iom141.html > > > -Original Message- > > From: James Bean [mailto:[EMAIL PROTECTED] > > Sent: Saturday, February 19, 2005 11:34 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] External relay triggered by Asterisk > > extension -question > > > > > > > > Has anyone every setup an external open/close relay, off > say a serial > > interface, and have an extension trigger the relay? > > > > Why I ask is I have a student accomodation where I am installing an > > asterisk box to supply phone services to the tenants, there > is already > > an intercom system in the main hallways that triggers the > downstairs > > door and gate using a standard relay open/close trip, so I > was hoping > > to get the linux box with asterisk to trip the same type of relay. > > > > Is there any door phones that are speaker driven only and sip based > > that anyone knows about as well? > > > > James > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/aster> isk-users To > > UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
On Sun, 20 Feb 2005, Duane wrote: > Anton Krall wrote: > > I think it would be your last suggestion.. When I pickup the phone I hear a > > tone, the sip phone box tone... Then I hit 9, no tones :) and enter the > > whole phone number and it starts to ring on the other side.. So no outside > > dialtone get heard ever.. I was wondering if it could be possible to make it > > so that after hitting 9.. The tone would change to something else letting > > the user know that they are dialing on an outside line. > > Yes, you can do this, stick a extension in your dial plan for 9, then > point that to app_disa... Or have the 9 dial an outside line and get the external dialtone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
On Sun, February 20, 2005 21:56, Peter Svensson said: > Or have the 9 dial an outside line and get the external dialtone. Which will only work if you're actually sending the call to an outside line... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "In the long run the pessimist may be proved right, but the optimist has a better time on the trip." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mandrake & CAPI
Title: Message All,I have been trying to get CAPI4Linux working on my machine and being frank am failing miserably! I am looking for any help available, I am real newbie (so please be gentle) and choose to run Mandrake 9.2 as it appears quite friendly (or so I thought!). I have been following the guidance found at http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI for the AVM card (actually I have a BT Speedway - apparently the same thing). I guess the best approach is to detail what I have done in tandem with the guidance? So here we go - Type - # modprobe capi Great! I get no response (which is expected!), so move to step 2 (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install) Guidance states 'Download and install your kernel sources' - I installed these as part of the original installation, so I'll ignore. I download and install the CAPI driver - # cd /usr/src # wget ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/fcpci-suse8.2-03.11.02.tar.gz # tar -xzvf fcpci-suse8.2-03.11.02.tar.gz # cd fritzGreat! Looking good! Guidance states modify the makefile in /usr/src/src.drv as follows - Replace - CARD_PATH = /lib/modules/`uname -r`/misc with - CARD_PATH = /lib/modules/$(uname -r)/kernel/drivers/isdn/avmb1 I am aware this chap is running Debian and I am running Mandrake, so after searching decided to modify the line as such - CARD_PATH = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1 Guidance states modify the KRNLINCL lines for the correct include path - KRNLINCL = /usr/src/kernel-headers-`uname -r`/include #KRNLINCL = /lib/modules/`uname -r`/build/include #KRNLINCL = /usr/src/linux/include And modify the lines as thus - DEFINES = -DMODULE -D__KERNEL__ -DNDEBUG \ -D__$(CARD)__ -DTARGET=\"$(CARD)\" CCFLAGS = -c $(DEFINES) -O2 -Wall -I $(KRNLINCL) With - DEFINES = -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG \ -D__$(CARD)__ -DTARGET=\"$(CARD)\" CCFLAGS = -c $(DEFINES) -march=i686 -O2 -Wall -I $(KRNLINCL) \ -include $(KRNLINCL)/linux/modversions.h Again aware of the Debian V's Mandrake configuration, I searched the web and found the following guidance for Mandrake (using the google translation feature - http://translate.google.com/translate?hl=en&sl=de&u=http://ixi.thepenguin.de/&prev=/search%3Fq%3Dcapi%2Bmandrake%26hl%3Den%26lr%3D%26rls%3DRNWE,RNWE:2004-35,RNWE:en ) And made the following changes to the makefile in /usr/src/src.drv as that seemed more appropriate and saved the file - KRNLINCL =/usr/src/linux/include DEFINES = Dmodule Dmodversions D__kernel __ Dndebug \D__$(card) __ Dtarget=\"$(card) \ " CCFLAGS = C $(defines) -march=i586 -O2 barrier i $(krnlincl) \include/usr/src/linux/include/linux/modversions.h Going back to the original Guidance (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install)I am instructed to modify the defs.h file in /usr/src/fritz/src.drv as follows - #if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) with #if LINUX_VERSION_CODE < KERNEL_VERSION(2, 4, 23) Great, I'm now ready to run the make command! Unfortunately the first couple of responses are as follows which to me looks very bad? And not sure what to do next? [EMAIL PROTECTED] src.drv]# makecc C Dmodule Dmodversions D__kernel__ DNDEBUG D Dtarget=\"\" -march=i586 -O2 barrier i /usr/src/linux/include include/usr/src/linux/include/linux/modversions.h main.c -o main.occ: C: No such file or directorycc: Dmodule: No such file or directorycc: Dmodversions: No such file or directorycc: D__kernel__: No such file or directorycc: DNDEBUG: No such file or directorycc: D: No such file or directorycc: Dtarget="": No such file or directorycc: barrier: No such file or directorycc: i: No such file or directorycc: include/usr/src/linux/include/linux/modversions.h: No such file or directory For completeness I Have included the makefile and defs.h files Makefile SOURCES = main.c driver.c tables.c queue.c lib.c tools.cOBJECTS = $(patsubst %.c,%.o,$(SOURCES)) LIBRARY = ../lib/$(CARD)-lib.o CARD_PATH = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1CS_PATH = /lib/modules/`uname -r`/pcmcia-external KRNLINCL = /usr/src/linux/include DEFINES = Dmodule Dmodversions D__kernel__ DNDEBUG \ D__$(CARD)__ Dtarget=\"$(CARD)\"CCFLAGS = C $(DEFINES) -march=i586 -O2 barrier i $(KRNLINCL) \ include/usr/src/linux/include/linux/modversions.hLDFLAGS = -r ifeq ($(CARD),fcpcmcia)CS_MOD = fcpcmcia_cs.oCS_SRC = fcpcmcia_cs.celseCS_MOD =CS_SRC =endif all: $(CARD).o $(LIBRARY) $(CS_MOD) install: $(CARD).o $(LIBRARY) $(CS_MOD) mkdir -p $(CARD_PATH) cp -f $(CARD).o $(CARD_PATH)ifeq ($(CARD),fcpcmcia) mkdir -p $(CS_PATH) cp -f $(CS_MOD) $(CS_PATH)endif clean:$(RM) $(OBJECTS) $(CARD).o $(CS_MOD) $(CARD).o: $(OBJECTS) $(LD) $(LDFLAGS) -o $@ $(OBJEC
[Asterisk-Users] FAX
I am using Underwood's fax system for fax on demand and it's very cool. I am planning to do the following and I would like to know if it's possible before putting my hands on it. For a specific application, I want to dialout thousands of numbers searching for fax machines. If somebody takes the call(voice), I would flag that number as bad in the DB. If it's a voice only answer machine, I would flag that number also as bad. But if it's a fax or an answer machine with fax, I would flag that number as valid fax number for future use. Is that possible? Thanks a lot, Isamar Maia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simulated dialtone like in other PBX
Easy as piece of cake. Remove ignorepat=>9 add: exten => 9,1,DISA(no-password|my_outbound_context) [my_outbound_context] exten => NXX, 1, blah-blah-blah All the Best! Sergey. Peter Svensson wrote: On Sun, 20 Feb 2005, Anton Krall wrote: Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? I'm not sure I understand your question. Do you want to be able to hit 9 and get a an outside line with dialtone? Just add an extension to do that. For isdn you need to enable overlap dialing. Or do you want Asterisk to provide a dialtone after the user have hit 9 as the first digit of a number? User the ignorepat option in the dialplan. Or do you want Asterisk to provide a _different_ dialtone after the user have hit 9 as the first digit of a number? This may be possible, but I think some hack may be needed. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
Hello, On Sun, 20 Feb 2005, Isamar Maia wrote: > For a specific application, > I want to dialout thousands of numbers searching for fax machines. > If somebody takes the call(voice), I would flag that number as bad in the > DB. If it's a voice only answer machine, I would flag that number also as > bad. But if it's a fax or an answer machine with fax, I would flag that > number as valid fax number for future use. > Is that possible? You are definitely in need of app_faxspam_harvest.so or am I wrong? Torsten Krueger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This is NUTS!!SOLVED
Title: Message So true. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed BradySent: Saturday, February 19, 2005 10:50 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] This is NUTS!!SOLVEDMakes you wonder about the future of CISCO doen't it? You are a potential customer trying every means possible to give them money, and they are making it difficult to do so. Most thriving businesses usually make it as convenient as possible for their customers to give them money. This reminds me of similar stories of Digital Equipment Corporation (DEC) before they fell on hard times.Ferguson, Michael wrote: Thanks everyone for your feedback, especially Mark. I now have the ALL the files I need. My order still stands for the $8.00 product from CISCO but the CP7960 dealer sent me all the files. Now I will move on to completeing the setup of the TFTP server. Thanks again -Original Message- From: Michael Loftis [mailto:[EMAIL PROTECTED]] Sent: Friday, February 18, 2005 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Ferguson, Michael Subject: Re: [Asterisk-Users] This is NUTS!! --On Friday, February 18, 2005 10:21 -0500 "Ferguson, Michael" <[EMAIL PROTECTED]> wrote: G'Day All; So I purchased a Cisco 7960 and am now trying to get it configured for *. No can do without the variuos files/images through a FTPF server. I configured the TFTP server on my RHES 3 box, now to get the required CISCO files. So I contacted CISCO to purchase the required maintenance contract so as to gain access to the download area for the files/images. -WHAT A FRUSTRATION!!- CISCO says, "Purchase it from your reseller/dealer." OK. So I call my reseller/dealer and he is having the most difficult time getting this $8.00 product, CON-SNT-CP7960, for me. It is just not worth the time and effort for him. So here I am, a week later, and no CP7960. It looks pretty though!! Can anyone recommend a speedier way to get this CON-SNT-CP7960 from CISCO Try contacting CDW, you'll need the phones serial number but they can probably help you out and get you the SMARTnet package. -- GPG/PGP --> 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
Actually, it was requested to me to build a fax number database. The real purpose is unknown. I am an IT guy, not marketing guy. Isamar On Sun, 20 Feb 2005, Torsten Krueger wrote: > Hello, > > On Sun, 20 Feb 2005, Isamar Maia wrote: > > > For a specific application, > > I want to dialout thousands of numbers searching for fax machines. > > If somebody takes the call(voice), I would flag that number as bad in the > > DB. If it's a voice only answer machine, I would flag that number also as > > bad. But if it's a fax or an answer machine with fax, I would flag that > > number as valid fax number for future use. > > Is that possible? > > You are definitely in need of app_faxspam_harvest.so or am I wrong? > > Torsten Krueger > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
On February 20, 2005 08:30 am, Isamar Maia wrote: > I want to dialout thousands of numbers searching for fax machines. You are an evil, evil man. Worse than the goddamned telemarketers, IMO. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
On February 20, 2005 01:41 am, Michael Giagnocavo wrote: > Well, sure, if you want to spend 8x the amount, yea, it's going to be a > much nicer setup. Show me a TDM404P for $100. Now show me a system with two of them working reliably and repeatably. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
Ok. I will be burned in fire.. :-) Or better.. I won't go to the heaven... Isamar On Sun, 20 Feb 2005, Andrew Kohlsmith wrote: > On February 20, 2005 08:30 am, Isamar Maia wrote: > > I want to dialout thousands of numbers searching for fax machines. > > You are an evil, evil man. Worse than the goddamned telemarketers, IMO. > > -A. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
Sorry, I understood the O.P. already had the hardware bought and installed and simply wanted to throw on an extra line. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Sunday, February 20, 2005 8:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines On February 20, 2005 01:41 am, Michael Giagnocavo wrote: > Well, sure, if you want to spend 8x the amount, yea, it's going to be a > much nicer setup. Show me a TDM404P for $100. Now show me a system with two of them working reliably and repeatably. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
On February 20, 2005 09:25 am, Michael Giagnocavo wrote: > Sorry, I understood the O.P. already had the hardware bought and installed > and simply wanted to throw on an extra line. You understood correctly; But again even a TDM401P is $133 on Digium's site. Considering you could probably get 60% of the price of your original TDM404P ($200 is 60% of $337), then you're either spending $133 for another TDM401P (and all the hassle of trying to get two to work in a system) or $600 (my $800 estimate - the $200 you got for your old equipmen)... $600 != 8x $133, and $800 != 8x $537. Anyway I think he's got his recommendations :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
On February 20, 2005 09:16 am, Isamar Maia wrote: > Ok. I will be burned in fire.. :-) > Or better.. I won't go to the heaven... heh. Either way I'm pretty sure you'll be on your own to write this kind of app. Personally I think you'd be FAR better off taking an electronic phonebook and SUBTRACTING any entries in there that didn't have "Fax" in the name and wardialing what was left. You'd certainly piss off a lot fewer people. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do ifyouneedmorethan4 C.O. lines
-Original Message- From: Andrew Kohlsmith > >On February 20, 2005 09:25 am, Michael Giagnocavo wrote: >> Sorry, I understood the O.P. already had the hardware bought and >installed >> and simply wanted to throw on an extra line. > >You understood correctly; But again even a TDM401P is $133 on Digium's site. >Considering you could probably get 60% of the price of your original >TDM404P >($200 is 60% of $337), then you're either spending $133 for another TDM401P >(and all the hassle of trying to get two to work in a system) or $600 (my >$800 estimate - the $200 you got for your old equipmen)... $600 != 8x $133, >and $800 != 8x $537. Yea, but if you buy a 2 port FXS for $75, that's $600/8. :) Not as elegant, but definitely another possibility (the o.p. seemed a tad dismayed he had to get all that hardware just to add on a line.). >Anyway I think he's got his recommendations :-) Yep! -Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR for callback
Some of my clients of hosted PBX service want to use it for callback when they cannot use the ATA. This is the scenario 1. Asterisk calls Party A at numA. 2. When A picks up the phone, he hears the announcement to enter the destination number, numB. He enters numB 3. Asterisk Dials numB and party A and B talk. This works well. However, there is no CDR generated. Is there a way I can force the CDR generation? If it helps, both the legs are on OH323 channels. Thanks, -- jt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peer registration interval - SOLUTION
This is what I tryied on last Tuesday. It ran fine until yesterday (4 days) then asterisk stopped re-registering again. A "sip reload" fixed the problem and asterisk now re-registers happily again. I'm just unsure for how long ... Stefan Gofferje wrote: Stefan Gofferje schrieb: Hi folks, I'm registered with sipgate, a German SIP provider. Configs works fine so far. Trouble is, after a while, it seems, my registration is dropped by sipgate. How do I tell * the interval for * registering with a provider? I suppose, the re-registration interval is to long... I finally found a solution. THe SER of Sipgate seem to dislike being qualified, so setting qualify=no solved this problem. I also set defaultexpirey=60, which is respected by Sipgate's SER and makes re-registration after change of dynamic IP a bit faster and more reliable. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External relay triggered by Asterisk extension-question
I just finnished a setup where I had the follwoing: 35 SIP phones. 2 TDM400 Cards with 8 FXO modules. An account by a VOIP provider for incoming/outgoing calls. Using private lines we connected 4 locations using DSL and T1. In each location we have a Bogen PCM 2000 Paging module (http://www.bogen.com/products/telephonepaging/) to do paging/night ring using either an FXO port (in the location where the * bos is located), or a SIP ATA. In each location we have a VikingElectronics C-2000 (http://www.vikingelectronics.com/) for door/gate opening, connected to either an FXO port, or a SIP ATA. Both the viking and the bogen allow you for relays to be hooked up. With the viking you can do using the door strike up to four relays, each configureable in different ways. BTW, Valcom (http://www.valcom.com/) also makes door openers. Just for the relay option, Viking has some better models. Hope this helps. On Sun, 20 Feb 2005 21:00:49 +1000, James Bean <[EMAIL PROTECTED]> wrote: > Very friggen cool, that you very much for the information it looks like > it will do the job nicely. > > What did you use in your extensions list to activate the relay? > > James > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk > > Sent: Sunday, 20 February 2005 6:24 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] External relay triggered by > > Asterisk extension-question > > > > Done something similar in a different application, but * > > should handle it -- > > > > In my case, I took a crystalfontz LCD, type 633, and used two > > of the four fan-outputs to drive two 12V relays. As a nice > > extra, you get temperature capabilities thrown in, so you can > > monitor your set-up. The LCD runs on serial, of course. > > > > As an alternative, you can use any of the many available > > relay boards -- $50 gets you this: > > http://www.phanderson.com/iom141.html > > > > > -Original Message- > > > From: James Bean [mailto:[EMAIL PROTECTED] > > > Sent: Saturday, February 19, 2005 11:34 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [Asterisk-Users] External relay triggered by Asterisk > > > extension -question > > > > > > > > > > > > Has anyone every setup an external open/close relay, off > > say a serial > > > interface, and have an extension trigger the relay? > > > > > > Why I ask is I have a student accomodation where I am installing an > > > asterisk box to supply phone services to the tenants, there > > is already > > > an intercom system in the main hallways that triggers the > > downstairs > > > door and gate using a standard relay open/close trip, so I > > was hoping > > > to get the linux box with asterisk to trip the same type of relay. > > > > > > Is there any door phones that are speaker driven only and sip based > > > that anyone knows about as well? > > > > > > James > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/aster> isk-users To > > > UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off
A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with "make samples", nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same trouble. I have several Uniden UIP200 phones and a single Grandstream BudgetTone 100. Any combination does the same thing. Calls started from within asterisk (*.call files, transfers, directory) work fine. I've tried all combinations of codecs, with no change. This is my first serious attempt with *, so don't be afraid to assume I'm a moron. Relevent config snippets and a "set verbose 100" and SIP DEBUG console dump follow. *** sip.conf *** [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes [1010] type=friend host=dynamic username=1010 secret=password context=default dtmfmode=rfc2833 <1011-1019 are all basically the same as 1010> *** extensions.conf *** [default] exten => 1010,1,Dial(SIP/1010,20,tr) exten => 1011,1,Dial(SIP/1011,20,tr) *** console dump of call, hold, unhold, hangup *** *** Asterisk on 192.168.200.0, phones on 192.168.201.0, *** connected by VPN, same thing happens when on one lan Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5 From: ;tag=9970b15421c8f59c To: Contact: Supported: replaces Call-ID: [EMAIL PROTECTED] CSeq: 22567 INVITE User-Agent: Grandstream BT100 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 354 v=0 =1019 0 8000 IN IP4 192.168.201.111 s=SIP Call c=IN IP4 192.168.201.111 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/8000 a=ptime:20 13 headers, 17 lines Using latest request as basis request Sending to 192.168.201.111 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 15 Found RTP audio format 99 Found RTP audio format 9 Peer audio RTP is at port 192.168.201.111:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Found description format iLBC Found description format G722 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5 From: ;tag=9970b15421c8f59c To: ;tag=as45319780 Call-ID: [EMAIL PROTECTED] CSeq: 22567 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="499f7907" Content-Length: 0 to 192.168.201.111:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '1019' asterisk*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5 From: ;tag=9970b15421c8f59c To: ;tag=as45319780 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 22567 ACK User-Agent: Grandstream BT100 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 11 headers, 0 lines asterisk*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828 From: ;tag=9970b15421c8f59c To: Contact: Supported: replaces Proxy-Authorization: DIGEST username="1019", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="499f7907", response="80ba81f6c2dc429b45c8bb6d57c9b7d6" Call-ID: [EMAIL PROTECTED] CSeq: 22568 INVITE User-Agent: Grandstream BT100 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 354 v=0 o=1019 1 8000 IN IP4 192.168.201.111 s=SIP Call c=IN IP4 192.168.201.111 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/8000 a=ptime:20 14 headers, 17 lines Using latest request as basis request Sending to 192.168.201.111 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 15 Found RTP audio format 99 Found RTP audio format 9 Peer audio RTP is at port 192.168.201.111:5004 Found description format PCMU Found description
RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
[EMAIL PROTECTED] wrote: > If you have a TDM card already, buying a T1, channelbank, > etc. to add a few lines is the stupidest thing I've heard of today. Not necessarily stupid, but certainly expensive. > Have you looked into buying some cheap multiport ATAs? 2 port > SIP/IAX2 ATA should be around $70-80? Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). > -Michael > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jim Van Meggelen > Sent: Saturday, February 19, 2005 8:39 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] A bit of a survey: What do do > if you needmorethan 4 C.O. lines > > Really? For five lines I need to buy all that hardware? > > Hmm. > > Well, I appreciate you taking the time to respond to my question. > > Regards, > > Jim. > > > [EMAIL PROTECTED] wrote: >> Digium tech support recommends going with a t1 card and a channel >> bank. This is by far the simplest, cheapest and cleanest solution >> that I know of. >> >> >> Jon. >> >> >> On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: >>> Folks, >>> >>> In light of all the troubles people report when running more than >>> one TDM400 card in a system, I wouldn't mind hearing what your >>> solution of choice would be when having to connect 5 or more analog >>> telco circuits to an Asterisk. >>> >>> I'll try and compile the answers together and get them into the >>> Wiki, as I figure this could be useful knowledge for the community. >>> >>> TIA, >>> >>> Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: If you have a TDM card already, buying a T1, channelbank, etc. to add a few lines is the stupidest thing I've heard of today. Not necessarily stupid, but certainly expensive. Have you looked into buying some cheap multiport ATAs? 2 port SIP/IAX2 ATA should be around $70-80? Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Saturday, February 19, 2005 8:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines Really? For five lines I need to buy all that hardware? Hmm. Well, I appreciate you taking the time to respond to my question. Regards, Jim. [EMAIL PROTECTED] wrote: Digium tech support recommends going with a t1 card and a channel bank. This is by far the simplest, cheapest and cleanest solution that I know of. Jon. On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. If you already need 5+ lines, and expect any growth, ask your telco to quote for a T1 with 6 (or 8) channels enabled. It might not be as expensive as you'd think, and you get all the advantages of a digital circuit, plus an easy expansion route. Plus you avoid the (possible) need for a channel bank. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do ifyouneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: > On February 20, 2005 09:25 am, Michael Giagnocavo wrote: >> Sorry, I understood the O.P. already had the hardware bought and >> installed and simply wanted to throw on an extra line. > > You understood correctly; Uh, nope. I've been unclear. It was a purely hypothetical situation. It could be a case of expansion, or a new system requiring more circuits than a single card supports. I think there's a gap. A channel bank tied into a T1 is kinda kludgy for such a small system. Technically sound, but kludgy. Any other system of that size wouldn't need all the integration gear. You'd just plug the lines in. I know, I know; this is Asterisk, and that means one has to be creative. Perhaps I already knew the answer before I asked the question . . . still, one can hope. > But again even a TDM401P is $133 on > Digium's site. Considering you could probably get 60% of the price of > your original TDM404P ($200 is 60% of $337), then you're either > spending $133 for another TDM401P (and all the hassle of trying to > get two to work in a system) or $600 (my $800 estimate - the $200 you > got for your old equipmen)... $600 != 8x $133, and $800 != 8x $537. Is there a place to buy brand new Adit600's with 5+ FXOs and a T1 card for $800? (I've looked on eBay, but that's not a reliable supply chain, and I have yet to see such a price for new equipment). Seems to me one is looking at more like $2000. > Anyway I think he's got his recommendations :-) Sure do! Thanks. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: > On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote: > >> [EMAIL PROTECTED] wrote: >>> If you have a TDM card already, buying a T1, channelbank, etc. to >>> add a few lines is the stupidest thing I've heard of today. >> >> Not necessarily stupid, but certainly expensive. >> >>> Have you looked into buying some cheap multiport ATAs? 2 port >>> SIP/IAX2 ATA should be around $70-80? >> >> Yep, that's a possibility, but it's rather more kludgy than I'd like. >> (heck, the channel bank and T1 is more kludgy than I'd like - an >> integrated card would be my preference). >> >> >>> -Michael >>> >>> -Original Message- >>> From: [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] On Behalf Of Jim >>> Van Meggelen Sent: Saturday, February 19, 2005 8:39 PM >>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >>> Subject: RE: [Asterisk-Users] A bit of a survey: What do do if you >>> needmorethan 4 C.O. lines >>> >>> Really? For five lines I need to buy all that hardware? >>> >>> Hmm. >>> >>> Well, I appreciate you taking the time to respond to my question. >>> >>> Regards, >>> >>> Jim. >>> >>> >>> [EMAIL PROTECTED] wrote: Digium tech support recommends going with a t1 card and a channel bank. This is by far the simplest, cheapest and cleanest solution that I know of. Jon. On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: > Folks, > > In light of all the troubles people report when running more than > one TDM400 card in a system, I wouldn't mind hearing what your > solution of choice would be when having to connect 5 or more > analog telco circuits to an Asterisk. > > I'll try and compile the answers together and get them into the > Wiki, as I figure this could be useful knowledge for the > community. > > TIA, > > Jim. >> > > If you already need 5+ lines, and expect any growth, ask your > telco to quote for a T1 with 6 (or 8) channels enabled. > > It might not be as expensive as you'd think, and you get all the > advantages of a digital circuit, plus an easy expansion route. I like the thinking; the challenge is often where in the world you are, and how much competition there is. Here in Ontario, T1's were generally priced such that fractional T1s hardly saved anything. There is more competition now, so prices are changing, but I still can't see frac T1 service competing with such a small number of analog circuits. I know there are places where such a thing could be had very competitively, so your advice is still good. > Plus you avoid the (possible) need for a channel bank. Agreed. Thanks kindly for the reply. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran Total Access MGCP Config?
I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line => aaln/1 line => aaln/2 The device is configured like this: MGCP Configuration | Standard MGCP 0.1 / NCS 1.0 MGCP Endpoint Config| MGC Address 192.168.1.253 | Local Address 192.168.2.2 | MGC UDP Port 2727 | Local UDP Port2427 | ADPCM Coding IETF (RTP) | RFC 2833 RTP Payload Type 94 | DSCP Signaling0 | DSCP RTP Traffic 0 | Advanced Config [+] I can ping between the devices fine. Doing an mgcp audit endpoint aaln/[EMAIL PROTECTED] gives retransmitting errors. tcpdump shows traffic over the wire. dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do if you need morethan 4 C.O. lines
Well, I appreciate everyone's input, and I'll give the matter some more thought. Just so no one stays up at night worrying, this is not a situation I am facing, it is simply a hypothetical scenario. As with so many things, there is always a trade-off between economy and functionality. The Adit 600 and T1 integration is certainly quality, but I have not been able find an economical way to do this (purchasing used equipment on eBay is fine for smaller deployments and lab gear, but not a very sound logistics strategy, and awfully difficult to explain to a customer). Again, thanks to everyone for their feedback. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: > Folks, > > In light of all the troubles people report when running more > than one TDM400 card in a system, I wouldn't mind hearing > what your solution of choice would be when having to connect > 5 or more analog telco circuits to an Asterisk. > > I'll try and compile the answers together and get them into > the Wiki, as I figure this could be useful knowledge for the > community. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
Jim Van Meggelen wrote: Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm Regards, -- Jason Becker Director & CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault
Hi, I'm trying to set up a fresh system for use with Asterisk. I've never installed or used Asterisk before, so I do not know much about it. I'm using Slackware Linux 10.1 and followed this guide: http://www.automated.it/guidetoasterisk.htm When I try to run asterisk though, at the point the guide suggests to try it, I get 'Segmentation fault'. Any idea what to do? Are there any known problems with Slackware and Asterisk? Thanks in advance, Julius ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom phone hint exten question
I haven't used it in a while, but I had to put subscribecontext=sip for the phone's (in your case the snom) sip entry. This seems like it has been removed from the wiki. Has it changed or is this incorrect? http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+phone+snom&diff=7 On Sat, 19 Feb 2005 21:36:04 +1000, James Bean <[EMAIL PROTECTED]> wrote: > Putting bt-karen in the destination of the snom doesn't work, i.e. > pushing the button the phone says no such destination. > > exten => 691,hint,SIP/bt-karen > exten => 691,1,SetMusicOnHold(random) > exten => 691,2,Dial(SIP/bt-karen,30,tr) > exten => 691,10,voicemail,u691 > > Is in the extensions.conf but in the snom I have destination as 691. > > In the sip.conf it is setup as > > [bt-karen] > type=friend > secret= > host=dynamic > callerid="Karen Colomb" <691> > defaultip=192.168.69.251 > dtmfmode=info > mailbox=691 > > Hope this helps. > > James -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do ifyouneedmorethan4 C.O. lines
On February 20, 2005 11:34 am, Jim Van Meggelen wrote: > Is there a place to buy brand new Adit600's with 5+ FXOs and a T1 card > for $800? (I've looked on eBay, but that's not a reliable supply chain, > and I have yet to see such a price for new equipment). Seems to me one > is looking at more like $2000. Why on earth would you buy a new one? Warranty? Screw that, buy two and have a spare ON HAND -- much better than relying on a courier to get a replacement to you overnight and your customer having the down time. Pricey for a one-off, sure, but if you've got two or three systems deployed that on-the-shelf spare is unbelievably cheap, especially since it's modular. This is something I recommend for anything as critical as a phone system. I'd suggest having a spare TDM400P with some modules onhand at all times, too. Ebay's as reliable as anything else for this stuff, in my experience. Unless you're going and installing one of these things a week or something, in which case I'm sure your price from CAC is going to be much better than $2k. I was negotiated a bulk buy of 24 (yes 24) entire Access Bank Is from an ebay vendor -- It feel through because I decided at the last moment that I really didn't need that many onhand and couldn't justify tying the money up in it. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
On February 20, 2005 11:44 am, Jim Van Meggelen wrote: > I like the thinking; the challenge is often where in the world you are, > and how much competition there is. Here in Ontario, T1's were generally > priced such that fractional T1s hardly saved anything. There is more > competition now, so prices are changing, but I still can't see frac T1 > service competing with such a small number of analog circuits. I know > there are places where such a thing could be had very competitively, so > your advice is still good. I think you'd be surprised. Even in Listowel a CT1 for POTS termination was on-par with having the individual analogue lines brought out. You'll pay a little more for the smartjack lease but it eliminates a lot of headaches. Hell the PRI here in cow-town Listowel was in-line with POTS until you included the D channel price of $500 -- The B chans were all $55/mo which is exactly what a business line costs. I imagine CT1 instead of PRI service would have been significantly cheaper, *AND* I wouldn't have to pay for all those extra DIDs. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if you need morethan 4 C.O. lines
On February 20, 2005 11:47 am, Jim Van Meggelen wrote: > As with so many things, there is always a trade-off between economy and > functionality. The Adit 600 and T1 integration is certainly quality, but > I have not been able find an economical way to do this (purchasing used > equipment on eBay is fine for smaller deployments and lab gear, but not > a very sound logistics strategy, and awfully difficult to explain to a > customer). There's nothing to explain to the customer. They want excellent customer service which you're providing on the equipment. Or skip it all entirely and lease it to them... I dunno, I've certainly never had any trouble. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
That app_disa is new to me... Is there a list of available apps? Im still quite new to asterisk but I guess you can find out which apps you have by using a show applications but my question would be more of how to make new apps or download/get new ones, is this possible? Also, is there a list of command that can be used in a dialplan or are they just apps like dial()? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Domingo, 20 de Febrero de 2005 04:48 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Anton Krall wrote: > I think it would be your last suggestion.. When I pickup the phone I > hear a tone, the sip phone box tone... Then I hit 9, no tones :) and > enter the whole phone number and it starts to ring on the other side.. > So no outside dialtone get heard ever.. I was wondering if it could be > possible to make it so that after hitting 9.. The tone would change to > something else letting the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "In the long run the pessimist may be proved right, but the optimist has a better time on the trip." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
from the console, "show modules" - "Yeah, we rocked the vote all right. Those little bastards betrayed us again." - Hunter S. Thompson on the 2004 election. On Sun, 20 Feb 2005, Anton Krall wrote: That app_disa is new to me... Is there a list of available apps? Im still quite new to asterisk but I guess you can find out which apps you have by using a show applications but my question would be more of how to make new apps or download/get new ones, is this possible? Also, is there a list of command that can be used in a dialplan or are they just apps like dial()? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Domingo, 20 de Febrero de 2005 04:48 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "In the long run the pessimist may be proved right, but the optimist has a better time on the trip." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
And go here: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands - "Yeah, we rocked the vote all right. Those little bastards betrayed us again." - Hunter S. Thompson on the 2004 election. On Sun, 20 Feb 2005, Anton Krall wrote: That app_disa is new to me... Is there a list of available apps? Im still quite new to asterisk but I guess you can find out which apps you have by using a show applications but my question would be more of how to make new apps or download/get new ones, is this possible? Also, is there a list of command that can be used in a dialplan or are they just apps like dial()? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Domingo, 20 de Febrero de 2005 04:48 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be possible to make it so that after hitting 9.. The tone would change to something else letting the user know that they are dialing on an outside line. Yes, you can do this, stick a extension in your dial plan for 9, then point that to app_disa... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers "In the long run the pessimist may be proved right, but the optimist has a better time on the trip." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
On Sun, 20 Feb 2005, Anton Krall wrote: > That app_disa is new to me... Is there a list of available apps? Im still > quite new to asterisk but I guess you can find out which apps you have by > using a show applications but my question would be more of how to make new > apps or download/get new ones, is this possible? "show applications" at the cli. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Prompts with no sound
I have a weird problem... very puzzling.. Yesterday I had sound problems with the voice prompts, I couldnt hear them, so I rebooted the system and voila, I was able to hear everything.. so I went to bad.. and I just woke up and tried the system again and its back!!! I dial the voicemail system and I cant hear the voice welcome.. I can hear any voice prompts Has anybody had this kind of problems? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FAX
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE I am using Underwood's fax system for fax on demand and it's very cool. I am planning to do the following and I would like to know if it's possible before putting my hands on it. For a specific application, I want to dialout thousands of numbers searching for fax machines. If somebody takes the call(voice), I would flag that number as bad in the DB. If it's a voice only answer machine, I would flag that number also as bad. But if it's a fax or an answer machine with fax, I would flag that number as valid fax number for future use. Is that possible? Thanks a lot, Isamar Maia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Ouch, Do you know how to use gdb, the Gnu Debugger? That will give you a clue as to where the segmentation fault is coming from. Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. 2. The dearth of information of value in your posting is amazing. I went to http://www.automated.it/guidetoasterisk.htm (a good start, good effort Mr. Powell.) As stated above, you life might be easier using FEDORA, not an endorsement of Red Hat, rather a plea for a unified Linux base (please don't say Debian, self-installing the micro-chip in my head was easier.) (it is the new Anti-AMD Intel Rantino chip for those interested.) 3. "I've never installed or used Asterisk before, so I do not know much about it." 1. What is your goal with installing Asterisk? 2. Do you have Digium or other hardware installed? 3. Are you running SIP/H323/MGCP? 4. Did you modify any files? 4. What was the last thing on the *CLI> screen before the seg fault? Come on Mr. Caesar throw us a bone here. All Hail, Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Schwartzenberg Sent: Sunday, February 20, 2005 11:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Segmentation fault Hi, I'm trying to set up a fresh system for use with Asterisk. I've never installed or used Asterisk before, so I do not know much about it. I'm using Slackware Linux 10.1 and followed this guide: http://www.automated.it/guidetoasterisk.htm When I try to run asterisk though, at the point the guide suggests to try it, I get 'Segmentation fault'. Any idea what to do? Are there any known problems with Slackware and Asterisk? Thanks in advance, Julius ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Race Vanderdecken wrote: Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. Why do you not follow Ann Landers simple adage, "Better to keep one's mouth shut and be thought a fool, than to open it and remove any doubt?" B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX
On Sun, 20 Feb 2005 23:16:00 +0900 (JST), Isamar Maia <[EMAIL PROTECTED]> wrote: > > Ok. I will be burned in fire.. :-) > Or better.. I won't go to the heaven... You are probably right. But in the the mean time, while you are here on earth, you will probably spend some time in the legal system too. Spam faxing is a punishable offense and enforced per incident. War dialing for fax machines fall under the same category. Spend a little time here before you get too far into the project. http://www.junkfax.org/index.html If you impede someone's ability to get the e911 system by clogging their lines that goes beyond illegal. Find another get rich quick scheme. -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote: > 1. Why are you running on Slackware? > Are you trying to prove a point or just enjoy being frustrated? > Open Source is like "Broad Spectrum Pesticide", it works but > your results may vary and you may end up killing your lawn. Got a problem with Slackware? It works *very* well with Asterisk. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Because I am more civilized?
By the way, it was Samuel Clemens's "fool..." quote, who stole it from
Mr. Lincoln, who stole if from Confucius (another educational Tyrant.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, February 20, 2005 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
Race Vanderdecken wrote:
> Good, then let me move on to the insults and ranting.
>
> 1. Why are you running on Slackware?
> Are you trying to prove a point or just enjoy being frustrated?
> Open Source is like "Broad Spectrum Pesticide", it works but
> your results may vary and you may end up killing your lawn.
>
Why do you not follow Ann Landers simple adage, "Better to keep one's
mouth shut and be thought a fool, than to open it and remove any doubt?"
B.
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[Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
Hello, I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same questionhow do I go from the RJ11 standard analog to the RJ45 on the TDM400P card? I'd appreciate any response. thx chuks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Brian Capouch wrote: Race Vanderdecken wrote: Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. Why do you not follow Ann Landers simple adage, "Better to keep one's mouth shut and be thought a fool, than to open it and remove any doubt?" Or maybe a double fool because he also disrespected Debian GNU/Linux in his reply. Is ignorance really bliss? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External relay triggered by Asteriskextension-question
Sorry, I didn't say I was using it with * -- just on a PC with a different app. I don't think it would be difficult to use something like lcdproc or even their test-app -- http://www.crystalfontz.com/software/633_WinTest/index.html (link to linux source at the bottom), and use agi to call the application. Basically, you got all the dots and all the connections. > -Original Message- > From: James Bean [mailto:[EMAIL PROTECTED] > Sent: Sunday, February 20, 2005 5:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] External relay triggered by > Asteriskextension-question > > > Very friggen cool, that you very much for the information it > looks like it will do the job nicely. > > What did you use in your extensions list to activate the relay? > > James > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf > Of Jay Milk > > Sent: Sunday, 20 February 2005 6:24 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] External relay triggered by > > Asterisk extension-question > > > > Done something similar in a different application, but * > > should handle it -- > > > > In my case, I took a crystalfontz LCD, type 633, and used two > > of the four fan-outputs to drive two 12V relays. As a nice > > extra, you get temperature capabilities thrown in, so you can > > monitor your set-up. The LCD runs on serial, of course. > > > > As an alternative, you can use any of the many available > > relay boards -- $50 gets you this: > > http://www.phanderson.com/iom141.html > > > > > -Original Message- > > > From: James Bean [mailto:[EMAIL PROTECTED] > > > Sent: Saturday, February 19, 2005 11:34 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [Asterisk-Users] External relay triggered by Asterisk > > > extension -question > > > > > > > > > > > > Has anyone every setup an external open/close relay, off > > say a serial > > > interface, and have an extension trigger the relay? > > > > > > Why I ask is I have a student accomodation where I am > installing an > > > asterisk box to supply phone services to the tenants, there > > is already > > > an intercom system in the main hallways that triggers the > > downstairs > > > door and gate using a standard relay open/close trip, so I > > was hoping > > > to get the linux box with asterisk to trip the same type of relay. > > > > > > Is there any door phones that are speaker driven only and > sip based > > > that anyone knows about as well? > > > > > > James > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/aster> isk-users To > > > UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Grasshopper, You have your first clue, the live test works. Do you understand how SIP works? During the INVITE sequence the Asterisk and the phone trade RTP CODEC information. RTP is the protocol that actually carries the sounds, SIP only does the handshaking for the call. A CODEC is what the RTP is carrying between the pones. If you do "sip debug" inside of the asterisk command line interface *CLI> sip debug Then you will see the SIP Messages and the Codec agreements. ... 16 headers, 13 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event -- Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) ... ... m=audio 19958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)" Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)" Above is the trace of my SNOM 200 -- see the "combined - 0xe (gsm|ulaw|alaw)"? The phone can do g729, but asterisk can't, so asterisk and the phone agree on a non-g729 codec, ulaw. Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk sound using ulaw at 8000hz. (again, I might be a little wrong on the extact details.) If the phones agree to use G729 then the playback won't work because you don't have a g729 license, $10 from Digium. Remember that asterisk is a third party to a conference and if your conference is using g729, then asterisk can't do that. In the sip.conf, Disallow=all Allow=gsm Allow=ulaw Allow=alaw This will force the phone and asterisk to speak gsm, ulaw or alaw. I had the same experience with no sound when I first connected a Cisco 7960, I could here other people, but not the prompts. Asterisk will allow G729 to pass through, but it will not allow G729 to originate and terminate without the license (I might be a little mistaken here...) I hope this helps. I have not use [EMAIL PROTECTED], it might be different. Let me know, Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 7:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Hi Race.. In this case, the asterisk|home comes preconfigured with some stuff different than the asterisk tar file. I check and the phone supports all mentioned codecs, I
Re: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
On Sun, 20 Feb 2005 [EMAIL PROTECTED] wrote: > > Hello, > I bought a TDM400P, and intended to use it with my analog phone, which is > RJ11 ofcourse. So, the question now, how do I plug in > my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since > it's an 11B card, I also intend to bring in an > analog line into the RJ45, so i am still left with the same questionhow > do I go from the RJ11 standard analog to the RJ45 on > the TDM400P card? I'd appreciate any response. Just plug the RJ11 into the socket - it will go in fine and work. This is a designed-in feature of the RJ series connectors, if I'm not mistaken. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
Push it with enough force, it will come in. On Sun, 20 Feb 2005 11:51:05 -0700, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hello, > I bought a TDM400P, and intended to use it with my analog phone, which is > RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the > TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I > also intend to bring in an analog line into the RJ45, so i am still left > with the same questionhow do I go from the RJ11 standard analog to the > RJ45 on the TDM400P card? I'd appreciate any response. > > thx > chuks > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
I have no problem with Slackware,
But when you are learning to drive a car you should first try a Chevy
with an automatic transmission first before strapping on a 6 speed
Ferrari.
Humor helps in teaching and getting a person to step out of a rut they
are having a problem in and gives them a chance to rethink what might be
going on.
Remember, my goal is to reduce the number of variables in the system.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, February 20, 2005 1:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
> 1. Why are you running on Slackware?
> Are you trying to prove a point or just enjoy being frustrated?
> Open Source is like "Broad Spectrum Pesticide", it works but
> your results may vary and you may end up killing your lawn.
Got a problem with Slackware? It works *very* well with Asterisk.
-A.
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RE: [Asterisk-Users] Simulated dialtone like in other PBX
Thx Sergey!! Ill give it a try -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Kuznetsov Sent: Domingo, 20 de Febrero de 2005 07:34 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Easy as piece of cake. Remove ignorepat=>9 add: exten => 9,1,DISA(no-password|my_outbound_context) [my_outbound_context] exten => NXX, 1, blah-blah-blah All the Best! Sergey. Peter Svensson wrote: >On Sun, 20 Feb 2005, Anton Krall wrote: > > > >>Im new to asterisk but is it possible to simulate a dialtone for >>example, in other PBX when you pick up the phone you can hear a >>certain dialup, which is the PBX dialtone, and when you hit 9, you can >>hear the PSTN dialtone, is this possible? >> >> > >I'm not sure I understand your question. > >Do you want to be able to hit 9 and get a an outside line with dialtone? >Just add an extension to do that. For isdn you need to enable overlap >dialing. > >Or do you want Asterisk to provide a dialtone after the user have hit 9 >as the first digit of a number? User the ignorepat option in the dialplan. > >Or do you want Asterisk to provide a _different_ dialtone after the >user have hit 9 as the first digit of a number? This may be possible, >but I think some hack may be needed. > >Peter > > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
Ok... I added the extension and here are the results:
-- Executing Wait("SIP/intruder-phone1-8613", "2") in new stack
-- Executing Answer("SIP/intruder-phone1-8613", "") in new stack
-- Executing Playback("SIP/intruder-phone1-8613", "vm-isunavail") in new
stack
-- Playing 'vm-isunavail' (language 'en')
On the sip phone I hear no prompts or recordings. :(
I tried rebooting the system, and weird, it worked once, and then, it
stopped working.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
This is getting hard.
So what do we have?
1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.
So let's go with the first principle: Eliminate the variables.
Do this:
Extensions.conf
[NoSound]
exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup
And see what happens.
I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?
Race "The Tyrant" Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Race.
Here are thre results of the tests:
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing)
Seems both can speak ulaw and ilbc but still, no prompt on the phone :(
Any ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Grasshopper,
You have your first clue, the live test works.
Do you understand how SIP works? During the INVITE sequence the Asterisk and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.
If you do "sip debug" inside of the asterisk command line interface
*CLI> sip debug
Then you will see the SIP Messages and the Codec agreements.
...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"
Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?
The phone can do g729, but asterisk can't, so asterisk and the phone agree
on a non-g729 codec, ulaw.
Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk
sound using ulaw at 8000hz. (again, I might be a little wrong on the extact
details.)
If the phones agree to use G729 then the playback won't work because you
don't have a g729 license, $10 from Digium.
Remember that asterisk is a third party to a conference and if your
conference is using g729, then asterisk can't do that.
In the sip.conf,
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
This will force the phone and asterisk to speak gsm, ulaw or alaw.
I had the same experience with no sound when I first connected a Cisco 79
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On Sun, 2005-02-20 at 13:51 -0500, Paul wrote: > Or maybe a double fool because he also disrespected Debian GNU/Linux in > his reply. *And* recommended Fedora, which makes it triple. I just dumped FC3 and replaced it with Debian because Fedora's kernels constantly gave me issues, e.g. with proprietary AVM kernel drivers which didn't even work. On the other hand, no probs whatsoever with Debian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Grasshopper, You have your first clue, the live test works. Do you understand how SIP works? During the INVITE sequence the Asterisk and the phone trade RTP CODEC information. RTP is the protocol that actually carries the sounds, SIP only does the handshaking for the call. A CODEC is what the RTP is carrying between the pones. If you do "sip debug" inside of the asterisk command line interface *CLI> sip debug Then you will see the SIP Messages and the Codec agreements. ... 16 headers, 13 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event -- Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) ... ... m=audio 19958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)" Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)" Above is the trace of my SNOM 200 -- see the "combined - 0xe (gsm|ulaw|alaw)"? The phone can do g729, but asterisk can't, so asterisk and the phone agree on a non-g729 codec, ulaw. Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk sound using ulaw at 8000hz. (again, I might be a little wrong on the extact details.) If the phones agree to use G729 then the playback won't work because you don't have a g729 license, $10 from Digium. Remember that aster
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
Ok Noload modems, alsa and oss... No errors... Is this ok? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Domingo, 20 de Febrero de 2005 01:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Grasshopper, You have your first clue, the live test works. Do you understand how SIP works? During the INVITE sequence the Asterisk and the phone trade RTP CODEC information. RTP is the protocol that actually carries the sounds, SIP only does the handshaking for the call. A CODEC is what the RTP is carrying between the pones. If you do "sip debug" inside of the asterisk command line interface *CLI> sip debug Then you will see the SIP Messages and the Codec agreements. ... 16 headers, 13 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event -- Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) ... ... m=audio 19958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)" Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)" Above is the trace of my SNOM 200 -- see the "combined - 0xe (gsm|ulaw|alaw)"? The phone can do g729, but asterisk can't, so asterisk and th
RE: [Asterisk-Users] A bit of a survey: What do do if you need morethan 4 C.O. lines
On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote: > Well, I appreciate everyone's input, and I'll give the matter some more > thought. > > Just so no one stays up at night worrying, this is not a situation I am > facing, it is simply a hypothetical scenario. > > As with so many things, there is always a trade-off between economy and > functionality. The Adit 600 and T1 integration is certainly quality, but > I have not been able find an economical way to do this (purchasing used > equipment on eBay is fine for smaller deployments and lab gear, but not > a very sound logistics strategy, and awfully difficult to explain to a > customer). This would be one of those cases where you keep a couple in stock and watch the ebay auctions when your stock goes low. You will find that your customers that are looking for the cheapest solutions possible will not baulk at used equipment. It is highly likely that they will price you against a used key system or pbx. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
Yes, running extra code/libraries/.so means more variables, which we are trying to eliminate. Really Anton I am stumped. Does anyone know? Do you have to have the gsm codec to hear the .gsm sound files. Is there an [EMAIL PROTECTED] mailing list? I found these: http://www.uninett.no/voip/asterisk.html No sound on SIP I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped" all sound out. I commented it out, and it was up and running on the sound. Now I just allow for the g.711 codec with disallow=all allow=ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Ok Noload modems, alsa and oss... No errors... Is this ok? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Domingo, 20 de Febrero de 2005 01:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Grasshopper, You have your first clue, the live test works. Do you understand how SIP works? During the INVITE sequence the Asterisk and the phone trade RTP CODEC information. RTP is the protocol that actually carries the sounds, SIP only does the handshaking for the call. A CODEC is what the RTP is carrying between the pones. If you do "sip debug" inside of the asterisk command line interface *CLI> sip debug Then you will see the SIP Messages and the Codec agreements. ... 16 headers, 13 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (non-NAT)
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 260
> From: "James Bean" <[EMAIL PROTECTED]>
> Has anyone every setup an external open/close relay, off say a serial
> interface, and have an extension trigger the relay?
The following will do the trick. Just add a 5vdc solid state relay
('cause you can't sink too much current out of the RS232C port).
Substitute "2", "4" or "6" in the code below to turn on either DTR, RTS
or both signals. "0" for off.
Change SWDEV in the lpswitch.h file to be the serial port you intend to
use for the relay. I'm using some optically isolated relays I found in
town for $5.00 Cdn. The box to put it in cost more than the relay.
There is a bunch of extra defines in the .h file that were needed for
the larger project this was part of. Just ignore them, they won't hurt.
Call this program from your dialplan, and voila.
Compile with cc -i lpon.c -o lpon
/*
* lpon.c Lineprinter ON
* *** test program only **
*
* (c) David Cook, 1994
*
* Set signlal lines on serial port to turn on 5vdc
* signal. Used for solid-state relay (low current
* draw on RS232C port) to switch high voltage/high
* current load for printer.
*
* Part of an intelligent print spooler to only power
* on/off high draw printer when required.
*
* Usage: lpon
* For example, lpon /dev/cua4 4 to set bit 3 on
* port /dev/cua4.
* "4" = 0100 or bit 3 which is DTR
* "2" = 0010 or bit 2 which is RTS
* "6" = 0110 or both DRT & RTS
*/
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include "lpswitch.h"
/* Main program. */
int main(int argc, char **argv)
{
struct termios port_config;
int fd;
int set_bits = 6;
/* Open monitor device. */
if ((fd = open(SWDEV, O_RDWR | O_NDELAY)) < 0) {
fprintf(stderr, "lpswtich: %s: %s\n", SWDEV, sys_errlist[errno]);
exit(1);}
cfmakeraw( &port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW, &port_config );
ioctl(fd, TIOCMSET, &set_bits );
sleep(5);
close(fd);
}
/* lpswitch.h
* include file for lpswitchd configuration
* (c) 1994, David Cook <[EMAIL PROTECTED]>
*/
#include
#define FILTERDEUG 0 /* filter app debug */
#define DAEMONDEBUG 0 /* daemon app debug */
#define VERSION "0.6" /* appl version number*/
#define LOCKF "/var/run/lpswitchd.pid" /* lock/PID file */
#define READYFILE "/tmp/lpready" /* printer ready file */
#define RQSTFILE"/tmp/lprequest" /* lprequest file */
#define LPDEV "/dev/lp0" /* physical lp device */
#define SWDEV "/dev/ttyC0"/* switch port-HW box */
#define SPEED B9600 /* port baud rate */
#define RESET B0 /* reset by 0 speed */
#define WARMUP 45 /* 45 sec warmup delay*/
#define IDLE1200/* 1200 seconds (20min)
idle delay */
#define XON 17 /* XON character */
#define XOFF19 /* XOFF character */
#define ABORTTIME 90 /* Max before abort */
dbc.
David Cook
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[Asterisk-Users] Recording of calls stopped - normal behaviour?
Hi all, I have call recording enabled via the Monitor command and it seems, the call stops being recorded after the call is transferred. Is this normal behavior? If so how can I continue recording of calls after they have been trasnferred ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?
Hi, I mistakenly posted this to Dev list I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN PBX and ISDN line - if power of Asterisks fails - will those card connect PBX directly to ISDN line ? If not are there any other simple switching devices, that would do this (in power fail it will connect ISDN PBX to ISDN lines directly) ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?
Hi, > I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN > PBX and ISDN line - if power of Asterisks fails - will those card connect > PBX directly to ISDN line ? No, you need a isdn failover switch > If not are there any other simple switching > devices, that would do this (in power fail it will connect ISDN PBX to ISDN > lines directly) ? Yes, klaus (author of bristuff) has/will have a solution for that. Hardware isdn failover switch. I don't know if I can reveal some details on this magic, so please contact him for further details Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: > Jim Van Meggelen wrote: > >> Yep, that's a possibility, but it's rather more kludgy than I'd like. >> (heck, the channel bank and T1 is more kludgy than I'd like - an >> integrated card would be my preference). > > I haven't followed this thread closely but have you looked into the > Voicetronix OpenSwitch cards? > > http://www.voicetronix.com.au/hda.htm I've looked at them, but never heard much about them. Is anyone using them? Can anyone give a comparison vs. the TDM400? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] possible attack, or just dumb log question?
I've got a strange situation that started yesterday -- I have a ton
of calls listed in the log for number = 18883629704
It initially looked like I was getting an incoming call on Zap/4 (LD
trunk) from 18883629704, which was going to an extension at Zap/2,
and then trying to dial out again back to the 18883629704 number
(the 'dial' application was called, with the argument
Zap/4/18883629704).
I found one reference to on Google to this number under the topic
"New ECM technique", describing what looks like some
kind
of attack on some unknown system (the domain is down, but it
was in the google cache)..
The outgoing attempts weren't working (apparently because they
were coming in on the same trunk that's used for LD outgoing),
but it was still disconcerting...
So I tried to block receiving any calls from 18883629704 in
the dialplan by giving them the congestion application, and also
blocking outgoing calls to it the same way, as
exten => s/3202594099,1,Congestion
exten => s/8883629704,1,Congestion
exten => s,1,Answer()
exten => s,2,NoOp(INCOMING call at ${DATETIME} from ${CALLERID}:
Name: ${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten => s,3,DigitTimeout(10)
exten => s,4,ResponseTimeout(20)
exten => s,5,Background(splash)
...
and
exten => 18883629704,1,Hangup()
in the [outgoing] context.
But I'm still getting these things, every 45 minutes or so, in pairs
about a minute or so apart. At least now they're not trying to dial
out, and the hangup seems to be working, but why is there all
this activity? And why am I getting the incoming digits that it's
trying to dial? It looks like they're not getting the congestion thing
at all?
I put the logging into verbose debug mode, and got the following,
which doesn't make a lot of sense. Shouldn't there be a log
entry for the Zap/4 (incoming trunk) call before it gets rung to the
Zap/2 (station) extension?
Thanks in advance for any help!
rj
2005-02-20 14:05:46 DEBUG[28229]: Monitor doohicky got event
Ring/Answered on channel 2
2005-02-20 14:05:46 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:05:46 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 1 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 3 on Zap/2-1
2005-02-20 14:05:48 DEBUG[28229]: DTMF digit: 6 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 2 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 9 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 7 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 0 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: DTMF digit: 4 on Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: Enabled echo cancellation on channel 2
2005-02-20 14:05:49 DEBUG[28229]: Launching 'Hangup'
2005-02-20 14:05:49 DEBUG[28229]: Spawn extension
(default,18883629704,1) exited non-zero on 'Zap/2-1'
2005-02-20 14:05:49 DEBUG[28229]: Hanging up channel 'Zap/2-1'
2005-02-20 14:05:49 DEBUG[28229]: zt_hangup(Zap/2-1)
2005-02-20 14:05:49 DEBUG[28229]: Hangup: channel: 2 index = 0, normal
= 16, callwait = -1, thirdcall = -1
2005-02-20 14:05:49 DEBUG[28229]: disabled echo cancellation on channel
2
2005-02-20 14:05:49 DEBUG[28229]: Set option TDD MODE, value: OFF(0) on
Zap/2-1
2005-02-20 14:05:49 DEBUG[28229]: Updated conferencing on 2, with 0
conference users
2005-02-20 14:05:49 DEBUG[28229]: Device 'Zap/2' changed to state '0'
2005-02-20 14:05:49 DEBUG[28229]: Device 'Zap/2' changed to state '0'
2005-02-20 14:05:50 DEBUG[28229]: Monitor doohicky got event Hook
Transition Complete on channel 2
2005-02-20 14:05:54 DEBUG[28229]: Monitor doohicky got event On hook on
channel 2
2005-02-20 14:05:54 DEBUG[28229]: disabled echo cancellation on channel
2
2005-02-20 14:06:06 DEBUG[28229]: Monitor doohicky got event
Ring/Answered on channel 2
2005-02-20 14:06:06 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:06:06 DEBUG[28229]: Device 'Zap/2' changed to state '2'
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 1 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:08 DEBUG[28229]: DTMF digit: 8 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 3 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 6 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 2 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 9 on Zap/2-1
2005-02-20 14:06:09 DEBUG[28229]: DTMF digit: 7 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: DTMF digit: 0 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: DTMF digit: 4 on Zap/2-1
2005-02-20 14:06:10 DEBUG[28229]: Enabled echo cancellation on channel 2
2005-02-20 14:06:10 DEBUG[28229]: Launching 'Hangup'
2005-02-20 14:06:10 DEBUG[28229]: Spawn extension
(default,1888362970
RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines
[EMAIL PROTECTED] wrote: > On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote: >> Well, I appreciate everyone's input, and I'll give the matter some >> more thought. >> >> Just so no one stays up at night worrying, this is not a situation I >> am facing, it is simply a hypothetical scenario. >> >> As with so many things, there is always a trade-off between economy >> and functionality. The Adit 600 and T1 integration is certainly >> quality, but I have not been able find an economical way to do this >> (purchasing used equipment on eBay is fine for smaller deployments >> and lab gear, but not a very sound logistics strategy, and awfully >> difficult to explain to a customer). > > This would be one of those cases where you keep a couple in > stock and watch the ebay auctions when your stock goes low. > You will find that your customers that are looking for the > cheapest solutions possible will not baulk at used equipment. > It is highly likely that they will price you against a used key > system or pbx. Certainly keeping spares in stock is good advice, and I don't mind using pre-owned equipment if it's solid stuff (which I know the adit is). I'm going to think about this some. As for price, that's always the challenge. Thing is, the lowest price does not always win. Still, being able to keep costs low is always going to be an advantage. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
Title: Message Just plug it in. The RJ11 is narrower than the RJ48, but has the exact same connection mechanism. it'll fit perfectly (the centre two pins are the contacts) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: February 20, 2005 1:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Digium TDM400P has RJ45 interface,how to connect to analog phone RJ11? Hello, I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same questionhow do I go from the RJ11 standard analog to the RJ45 on the TDM400P card? I'd appreciate any response. thx chuks --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran Total Access MGCP Config?
Dave Weis wrote: I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line => aaln/1 line => aaln/2 Check that the name adtran can be resolved by your DNS or /etc/hosts. Otherwise just put in the IP address. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom phone hint exten question
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jon Radon
> Sent: Monday, 21 February 2005 2:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Snom phone hint exten question
>
> I haven't used it in a while, but I had to put
> subscribecontext=sip for the phone's (in your case the snom)
> sip entry.
>
> This seems like it has been removed from the wiki. Has it
> changed or is this incorrect?
>
> http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+ph
> one+snom&diff=7
>
>
> On Sat, 19 Feb 2005 21:36:04 +1000, James Bean
> <[EMAIL PROTECTED]> wrote:
> > Putting bt-karen in the destination of the snom doesn't work, i.e.
> > pushing the button the phone says no such destination.
> >
> > exten => 691,hint,SIP/bt-karen
> > exten => 691,1,SetMusicOnHold(random)
> > exten => 691,2,Dial(SIP/bt-karen,30,tr) exten =>
> 691,10,voicemail,u691
> >
> > Is in the extensions.conf but in the snom I have destination as 691.
> >
> > In the sip.conf it is setup as
> >
> > [bt-karen]
> > type=friend
> > secret=
> > host=dynamic
> > callerid="Karen Colomb" <691>
> > defaultip=192.168.69.251
> > dtmfmode=info
> > mailbox=691
> >
> > Hope this helps.
> >
> > James
>
>
> --
> Is it something someone said, was it something someone said?
> ___
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>
>
Thanks for the link, it had some very userful information in it,
unforunately the lights on my snom are still dead as a door nail.
Ok the snom phone has one of its LED's set to Destination 691 (it
changes that into the sip address and it dials the extension when I hit
the button on the snom no problems, and the led works)
Does anyone know where I have gone wrong.
Configurations I have enabled are voicemail and call parking.
My sip.conf is
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = all
allow = ilbc
allow = alaw
allow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip
[snom-james]
type=friend
secret=
host=dynamic
callerid="James Bean" <690>
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690
[bt-karen]
type=friend
secret=
host=dynamic
callerid="Karen Colomb" <691>
defaultip=192.168.69.251
dtmfmode=info
mailbox=691
My extensions.conf is
[pstn]
exten => s,hint,SIP/bt-karen
exten => s,1,SetMusicOnHold(random)
exten => s,2,Dial(SIP/snom-james&SIP/bt-karen,45,t)
exten => s,4,VoiceMail(u690)
exten => s,5,Hangup
[internal]
exten => i,1,Playback(invalid)
exten => i,2,Hangup
exten => t,1,Hangup
exten => 098,1,WaitMusicOnHold(45)
exten => 099,1,Echo ;simple echo test when you dial 099 on your
phone
exten => 1690,1,VoicemailMain,s690
exten => 1691,1,VoicemailMain,s691
[outgoing]
exten => _9X.,hint,SIP/bt-karen
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten => _9X.,2,Congestion()
exten => _9X.,3,Hangup
[sip]
exten => 690,hint,SIP/snom-james
exten => 690,1,SetMusicOnHold(random)
exten => 690,2,Dial(SIP/snom-james,30,Ttr)
exten => 690,3,Voicemail,u690
exten => 690,103,Voicemail,b690
exten => 691,hint,SIP/bt-karen
exten => 691,1,SetMusicOnHold(random)
exten => 691,2,Dial(SIP/bt-karen,30,Ttr)
exten => 691,3,Voicemail,u691
exten => 691,103,Voicemail,b691
include => internal
include => outgoing
include => parkedcalls
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RE: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
[EMAIL PROTECTED] wrote:
> I have no problem with Slackware,
Me neither. I learned Linux with Slack. Found it to be extremely
friendly. And that was 10 years ago. Last time I chacked, it was still
friendly (and not at all GUI, unless you want it served that way)
> But when you are learning to drive a car you should first try
> a Chevy with an automatic transmission first before strapping
> on a 6 speed Ferrari.
Popular opinion holds that people who learn to drive standard first
generally end up being better drivers. And why wouldn't you want to
learn on a Ferrari since you can get one for free!?!
> Humor helps in teaching and getting a person to step out of a
> rut they are having a problem in and gives them a chance to
> rethink what might be going on.
Ya, but humour should be dispensed carefully, lest offence be given.
> Remember, my goal is to reduce the number of variables in the system.
The problem I see with Fedora is that you can install it successfully
without learning anything about Linux. Slackware is rather good for
learning Linux, because it is friendly and helpful, but still expects
you to make the decisions. I'd argue that a familiarity with the shell
is going to be essential for even a basic Asterisk install. It's not a
pre-qualifier so much as an essential skill.
LOL! You're just bored and are trolling for a holy war, eh? Well, I
guess we gotta shake off these Febraury blah's somehow.
GENTOO IS FOR WANNABE NEWBIES!!! (that oughta stir things up)
--
Jim Van Meggelen
[EMAIL PROTECTED]
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Sunday, February 20, 2005 1:39 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [Asterisk-Users] Segmentation fault {Writer
> given gnu-lashing}
>
> On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
>> 1. Why are you running on Slackware?
>> Are you trying to prove a point or just enjoy being frustrated?
>> Open Source is like "Broad Spectrum Pesticide", it works but your
>> results may vary and you may end up killing your lawn.
>
> Got a problem with Slackware? It works *very* well with Asterisk.
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
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[Asterisk-Users] help with @home
Title: Message just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wiki down?
It seems to me wiki downtime is somehow regular. Is this the fact? If so, should it be moved? roy On Feb 19, 2005, at 10:02 PM, James H. Thompson wrote: Wiki is back up. Between comment SPAM storms, over eager robots ignoring robots.txt, and mysql issues, it has been an interesting week. Jim James H. Thompson [EMAIL PROTECTED] [EMAIL PROTECTED] - Original Message - From: Roy Sigurd Karlsbakk To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, February 19, 2005 8:13 AM Subject: [Asterisk-Users] wiki down? hi is the wiki down again? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4 C.O. lines
Yesterday, I've checked tariffs from Bell Canada, For Full voice T1 it was costs around $1000 + tax. $216 - is access fee, $34 per channel. You can get the PRIs from Allstream with 3 years commitment ~$600 per month. Andrew Kohlsmith wrote: On February 20, 2005 11:44 am, Jim Van Meggelen wrote: I like the thinking; the challenge is often where in the world you are, and how much competition there is. Here in Ontario, T1's were generally priced such that fractional T1s hardly saved anything. There is more competition now, so prices are changing, but I still can't see frac T1 service competing with such a small number of analog circuits. I know there are places where such a thing could be had very competitively, so your advice is still good. I think you'd be surprised. Even in Listowel a CT1 for POTS termination was on-par with having the individual analogue lines brought out. You'll pay a little more for the smartjack lease but it eliminates a lot of headaches. Hell the PRI here in cow-town Listowel was in-line with POTS until you included the D channel price of $500 -- The B chans were all $55/mo which is exactly what a business line costs. I imagine CT1 instead of PRI service would have been significantly cheaper, *AND* I wouldn't have to pay for all those extra DIDs. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- All the Best! Sergey. = Sergey Kuznetsov President/CEO High Intellectual Technologies, Inc. Web: http://www.hitcalls.com E-mail: [EMAIL PROTECTED] Business phone: (416) 548-9700 Mobile phone: (647) 287-8448 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting if a call is active on chan_sip before trying INVITE? (was Sip question - allow only 1 incoming call to sip phone)
Disable the call waiting feature in the phone, so it will signal "486 - Busy here" to additionally incoming calls. Is it possible to test if a call to SIP/xxx is in place before dialling out? This could help a lot to centralize administation of whether or not to use call waiting instead of configuring the ATAs. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conecting to asterisk server through NAT using IAX
Hello, I have asterisk setup with Broadvoice. It works great as PBX and Outgoing calling server for all local clients withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my asterisk server. 5060-5063 1-2 5036 4569 It works ok with broadvoice, but clinets cannot connect to the server. This is my iax.conf file [general] port=5036 tos=lowdelay jitterbuffer=no disallow=all allow=ulaw allow=ilbc allow=gsm allow=adpcm allow=alaw register => xxx:[EMAIL PROTECTED] [guest] type=user context=abcxyz auth=none [voicepulse-in-01] ; <-- Name must be [voicepulse-in-01] type=user context=voicepulse-incoming ; <-- Should match the context you ; are using in extensions.conf auth=rsa inkeys=voicepulse01 [tester] type=friend context=sip auth=plaintext secret=secrwt host=dynamic allow=all nat =1 Clients cannot connect to asterisk. WHY??? Am I doing something wrong? Please help. Thanks Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Thanks a lot for your message. Race Vanderdecken schreef: Ouch, Do you know how to use gdb, the Gnu Debugger? That will give you a clue as to where the segmentation fault is coming from. No, I once used it being instructed exactly by a developer to solve a problem in Dosemu, but I never did anything else with it. I understand that I need to recompile Asterisk with debugging support. Could you give me some pointers on what to do next? Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. I'm using a pretty old system and I have good experiences with Slackware on other systems. Here are the specs of the system I'm using: IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB. 2. The dearth of information of value in your posting is amazing. I went to http://www.automated.it/guidetoasterisk.htm (a good start, good effort Mr. Powell.) As stated above, you life might be easier using FEDORA, not an endorsement of Red Hat, rather a plea for a unified Linux base (please don't say Debian, self-installing the micro-chip in my head was easier.) (it is the new Anti-AMD Intel Rantino chip for those interested.) I've never used Fedora on older systems, but I thought it wouldn't run very well on the system I'm using. (Good thing you don't have an Anti-IBM chip ;) 3. "I've never installed or used Asterisk before, so I do not know much about it." 1. What is your goal with installing Asterisk? We have about 8 telephones that use the plain telephone system to call each other and externally. Some of them are analog and others are digital (ISDN). I've also still got the old ISDN card from before we had ADSL. (Eicon Diva 2.01 ISA, seems to work with the hisax module.) Since I read that Asterisk worked with any ISDN adapter that was supported by ISDN4Linux, I thought it might be possible to hook it up in such a way that the phones could call the Asterisk system and that Asterisk would forward the call to a computer (and maybe even over the internet). Also the other way around would be neat. 2. Do you have Digium or other hardware installed? No. Only the ISDN adapter. 3. Are you running SIP/H323/MGCP? No. I've experimented with SIP before, but only with a softphone, using an account from SIPPhone.com. It would be nice if I could call my Asterisk system using SIP! 4. Did you modify any files? None from Asterisk. 4. What was the last thing on the *CLI> screen before the seg fault? The command to run Asterisk. It immediatly gives the error when I try to run it. Is Asterisk able to do, what I thought it would do or am I just messing? Come on Mr. Caesar throw us a bone here. All Hail, Race "The Tyrant" Vanderdecken Thanks again, Julius ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant"
It is weird.. I did a full asterisk reinstall... (no asterisk at home now)... And well Problem persists but this is weird, when it happens, I reboot the machine, starts working again and sometimes sound stops, sometimes it doesnt... This machine seems to have an attitude :) Last reboot one of the x100p cards complained during the modprobe wcfxo... :) then it didnt :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 02:07 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Yes, running extra code/libraries/.so means more variables, which we are trying to eliminate. Really Anton I am stumped. Does anyone know? Do you have to have the gsm codec to hear the .gsm sound files. Is there an [EMAIL PROTECTED] mailing list? I found these: http://www.uninett.no/voip/asterisk.html No sound on SIP I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped" all sound out. I commented it out, and it was up and running on the sound. Now I just allow for the g.711 codec with disallow=all allow=ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Ok Noload modems, alsa and oss... No errors... Is this ok? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Domingo, 20 de Febrero de 2005 01:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing
Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On Sun, 2005-02-20 at 22:59 +0100, Julius Schwartzenberg wrote: > I'm using a pretty old system and I have good experiences with Slackware > on other systems. Here are the specs of the system I'm using: > IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB. Here is your trouble. The Cyrix chip is what is the newer Via chipsets are based on. It isn't a real pentium chipset and needs to get tuned down via the CFLAGS to 586 or lower. You will probably hit the limits of that machine really quickly. You may want to find a slightly better machine for testing. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ; Possible heat problem
Okay, now you are getting off track. Hold old is the motherboard? How big is the case? How big is the power supply? If it is a smaller case and server then sometimes heat can be an issue when you are on the threshold of the temperature limit. Things will work mysteriously and then not work. Make sure all your cards in their slots tightly and screwed down. Try running with the case cover off the server. If it then runs fine, you have an overheating problem. As an example my eth1 PCI network card was failing intermittently. Turns out it was really the Sound Blaster card that was loose and causing problems. Took the sound blaster out and eth1 is solid as a rock. Race Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 5:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" It is weird.. I did a full asterisk reinstall... (no asterisk at home now)... And well Problem persists but this is weird, when it happens, I reboot the machine, starts working again and sometimes sound stops, sometimes it doesnt... This machine seems to have an attitude :) Last reboot one of the x100p cards complained during the modprobe wcfxo... :) then it didnt :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 02:07 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Yes, running extra code/libraries/.so means more variables, which we are trying to eliminate. Really Anton I am stumped. Does anyone know? Do you have to have the gsm codec to hear the .gsm sound files. Is there an [EMAIL PROTECTED] mailing list? I found these: http://www.uninett.no/voip/asterisk.html No sound on SIP I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped" all sound out. I commented it out, and it was up and running on the sound. Now I just allow for the g.711 codec with disallow=all allow=ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Ok Noload modems, alsa and oss... No errors... Is this ok? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Domingo, 20 de Febrero de 2005 01:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Race. Here are thre results of the tests: Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw
Re: [Asterisk-Users] Detecting if a call is active on chan_sip before trying INVITE? (was Sip question - allow only 1 incoming call to sip phone)
Roy Sigurd Karlsbakk wrote: Is it possible to test if a call to SIP/xxx is in place before dialling out? This could help a lot to centralize administation of whether or not to use call waiting instead of configuring the ATAs. app_groupcount can be used to provide call counting in any fashion you desire. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sparc hardware, Linux and X100P REVISITED
I was studying the asterisk-users list archives to learn if anyone has had success with an X100P on a sparc. I noticed some postings on the subject. I am wondering if anyone has learned anything new? I have an Ultra-60 running Gentoo with 2.6.10 and udev. I built * 1.0.5 and have been enjoying various SIP configurations, with 2 sipura phones and 2 UIP200 phones (got them working!) in my home, bridging in FWD and now Voiptalk too. I bought 2 X100P clones via ebay, and put one in my U60. I can see it with lspci: 0001:00:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I successfully built zaptel, and can modprobe zaptel and wcfxo: # lsmod Module Size Used by wcfxo 14680 0 zaptel195424 1 wcfxo crc_ccitt 2752 1 zaptel However I can't ztcfg with any success: # ztcfg -v Zaptel Configuration == 1 channels configured. ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? In fact dmesg never shows wcfxo completely setting up the card: # dmesg Zapata Telephony Interface Registered on major 196 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone PCI Target abort PCI Target abort PCI Target abort I have to think that the PCI driver does not get along with the wcfxo driver for the X100P clone. Also, strange things happen while the driver is loaded too... consoles dropping, etc. As for pure SIP related functions, * 1.0.5 on sparc has performed very well, with all functions (moh, vm, xfer, call park, etc) working admirably. Has anyone been able to get any farther than I have with the X100P on a sparc? Rob __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
Title: Message Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser Sent: Sunday, February 20, 2005 7:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLinc www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wiki down?
On Sun, 20 Feb 2005 22:45:42 +0100, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote: > It seems to me wiki downtime is somehow regular. > Is this the fact? > If so, should it be moved? Just to add some balance to this threadJim and colleagues, thanks for hosting the Wiki. You should take it as a compliment that when it's down occasionally, so many people notice. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with @home
Kurt Fankhauser wrote: just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Are you logged into the console while your testing the dialing in? What messages are you seeing? If asterisk is already running in the background, do a "asterisk -r" before you start to dial in. If there is some other interface in the @home distribution for monitoring asterisk, you'll have to say what app you're using and what you're seeing. At any rate, without log, error, or console messages there's not alot we can do for you. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S ISDN card on *@home
It seems so simple, but I'm having no luck installing a HFC-s ISDN BRI card on [EMAIL PROTECTED] 0.5. I probably have to install BRI-stuff from Junghanns.net but that also downloads and installs another copy of * from Digium. I'm not sure if zaphfc has to be installed *before* Asterisk or if it's OK to do this afterwards. I've seen this question before, but: Anyone successfully installed a HFC-s card on [EMAIL PROTECTED] Please post the steps you had to take. I'm sure quite some list-members are interested! With kind regards Erwin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ; Possible heat problem
Just to be sure.. I checked the cards... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 04:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" ;Possible heat problem Okay, now you are getting off track. Hold old is the motherboard? How big is the case? How big is the power supply? If it is a smaller case and server then sometimes heat can be an issue when you are on the threshold of the temperature limit. Things will work mysteriously and then not work. Make sure all your cards in their slots tightly and screwed down. Try running with the case cover off the server. If it then runs fine, you have an overheating problem. As an example my eth1 PCI network card was failing intermittently. Turns out it was really the Sound Blaster card that was loose and causing problems. Took the sound blaster out and eth1 is solid as a rock. Race Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 5:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" It is weird.. I did a full asterisk reinstall... (no asterisk at home now)... And well Problem persists but this is weird, when it happens, I reboot the machine, starts working again and sometimes sound stops, sometimes it doesnt... This machine seems to have an attitude :) Last reboot one of the x100p cards complained during the modprobe wcfxo... :) then it didnt :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 02:07 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Yes, running extra code/libraries/.so means more variables, which we are trying to eliminate. Really Anton I am stumped. Does anyone know? Do you have to have the gsm codec to hear the .gsm sound files. Is there an [EMAIL PROTECTED] mailing list? I found these: http://www.uninett.no/voip/asterisk.html No sound on SIP I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped" all sound out. I commented it out, and it was up and running on the sound. Now I just allow for the g.711 codec with disallow=all allow=ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" Ok Noload modems, alsa and oss... No errors... Is this ok? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Domingo, 20 de Febrero de 2005 01:18 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" I dont know if it has something to do but I see 2 mpg123 processes running: 3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Everytime I start asterisk.. Also, if I enable alsa I get this error: [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459 load_modules: Loading module chan_alsa.so failed! [EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe And asterisk quits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Domingo, 20 de Febrero de 2005 12:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" This is getting hard. So what do we have? 1. The Asterisk server and the phones are using good CODECS. 2. Sound is moving from phone to phone. 3. Sound from the prompts is not playing back to the phones. So let's go with the first principle: Eliminate the variables. Do this: Extensions.conf [NoSound] exten => 222,1,Wait(2) exten => 222,2,Answer exten => 222,3,Playback(vm-isunavail) exten => 222,4,Hangup And see what happens. I might be missing something. Anyone know how .gsm files are translated to ulaw/alaw in asterisk? Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 10:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Di
RE: [Asterisk-Users] help with @home
Title: Message I'll buy a IP phone tomarrow so i can do that -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Sunday, February 20, 2005 2:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] help with @home Can you work through a process of elimination if you record the file using an internal extension by dialing *77 and seeing if that works? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent: Sunday, February 20, 2005 7:42 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] help with @home just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with @home
I think the box is answering calls but I don't think the digital receptionist is working properly. Kurt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Sunday, February 20, 2005 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help with @home Kurt Fankhauser wrote: > just reinstalled @home and i have a one of those 100 cards, anyways when > i call from the pstn the box picks up but i hear nothing, then it clicks > a couple times, then nothing again, i am trying to get the digital > receptionist to work but it won't save my wav file to the @home box and > all the radio buttons under incoming calls are greyed out. the greyed > out thing seems to be my biggest problem right now, also do you have to > use a ip phone to record your greeting because this wav file stuff isn't > working. Are you logged into the console while your testing the dialing in? What messages are you seeing? If asterisk is already running in the background, do a "asterisk -r" before you start to dial in. If there is some other interface in the @home distribution for monitoring asterisk, you'll have to say what app you're using and what you're seeing. At any rate, without log, error, or console messages there's not alot we can do for you. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with @home
Message> also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. I didn't try uploading. You can just setup a SIP softphone and dial *77 when looking at the menu you want to record in the GUI. Regards, Erwin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with @home
Message> I'll buy a IP phone tomarrow so i can do that No need: http://www.xten.net/index.php?menu=products&smenu=download Regards, Erwin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
