Re: [Asterisk-Users] Stuck On ISDN
Olle E. Johansson wrote: snip Everyone points to capi and, back to the start of my reply, it seems expensive for personal use... A passive AVM Fritz card is somewhere around € 100 /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One way voice through NAT
I'm connecting and can place calls to and from my SIP phone that is behind a firewall, can hear audio from the SIP on the PSTN line but can't hear audio on the SIP phone from the PSTN line. Anyone else experience this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] frames/packet
Noticed that I can adjust the number if frames/packet on the GrandStream phone. Can * do the same? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN server from Vovida
Not sure if it's alright to talk about this here???
compiled the STUN server from Vovida on RedHat 7.3. Looks simple to
configure. It isn't starting...it tries to for a long time and then just
craps out. Here is my config:/etc/sysconfig/stund
#!/bin/echo Not to execute.
# Path to stund
STUND=/usr/sbin/stund
# Set the required args for STUND
STUNDPRIMARYHOSTNAME=208.x.x.x
# The hostname where another stund server is running on port and
alternate
# port.
STUNDALTERNATEHOSTNAME=127.0.0.1
# The primary response port to user
STUNDPRIMARYPORT=3478
# The alternate port to use
STUNDALTERNATEPORT=3479
STUNDARGS="-h ${STUNDPRIMARYHOSTNAME} \
-p ${STUNDPRIMARYPORT} \
-a ${STUNDALTERNATEHOSTNAME} \
-o ${STUNDALTERNATEPORT}"
Any ideas? Any suggestion for another STUN server?
--
Paul
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Re: [Asterisk-Users] DTMF Tones During Call
On Wed, 3 Sep 2003, Jay Tyndall wrote: > I am receiving calls via a Netjet-S card on asterisk, and I notice that > whenever I am talkimng to someone, if their voice is loud enough, sometimes > asterisk generates a DTMF Tone as they speak. that is played to me. (Caller > doesn't hear it). > > Any ideas how to stop this? Check the list archive. This has been discussed a number of times. You'll need patches (should be archived as well.) Regards, Jac -- Jac KersingTechnical Consultant The-Box Development [EMAIL PROTECTED] http://www.the-box.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN
On Wed, 3 Sep 2003, Jay Tyndall wrote:
> I am using a Netjet-s ISDN Card, and am having some trouble dialling out
> (Incoming Works Fine).
...
> I get the following when diallingout:
> -- Starting simple switch on 'Zap/2-1'
>-- Executing Dial("Zap/2-1", "Modem/ttyI0/04||Ttm") in new
Check the line 'stripmsd=1'. If the number to be dialed does not need to
have the most significant digit stripped this line needs to be
commented/removed in order to dial a valid number. (with stripmsd active
the number dialed in your case would be 4)
Regards,
Jac
--
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[EMAIL PROTECTED] http://www.the-box.com
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[Asterisk-Users] IXJ card doesn't want to dial out (see previous thread, asterisk won't answer pstn ring)
Hi all,
Currently trying to get asterisk to dial out with an Internet Line Jack card,
however, it does not use the pots line, only on the line it dials out of. This
is similar to the previous thread/posting "Asterisk won't answer pstn ring",
but I didn't find any follow up to get it working.
My asterisk setup is like this:
iptelephony:/etc/asterisk# cat phone.conf | grep -v \;
[interfaces]
format=slinear
echocancel=medium
silencesupression=yes
context=local
mode=dialtone
device => /dev/phone0
mode=fxo
device => /dev/phone1
phone0 operates fine.
With regards to the following piece of code, I do not see the error message.
if (mode == MODE_FXO) {
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN))
ast_log(LOG_DEBUG, "Unable to set port to PSTN\n");
} else {
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
ast_log(LOG_DEBUG, "Unable to set port to POTS\n");
}
Using Debian 3, updated, with the ixj driver version 1.2.1.
Any suggestions would be appreciated.
Thanks,
Andrew Griffiths
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[Asterisk-Users] Designing a lab for a telecommuncations course using Asterisk
Hello all, I have been asked to help in the design of a lab for a telecommunications technology course (third year students) to teach VoIP technologies. Here is a cut n' paste of an email I received from one of my instructors "A very important consideration is the numbering plan and if telephone domains (areas) can be established. In principle I suppose so. My idea is to have local, interareas, and finally Internet voice calls. The lab would include BGP, OSPF or ISIS, and maybe MPLS. That's what is ticking in my mind. Who knows at the end, really. When I get to design labs I become a little too creative." Here is a little bit about how the lab is setup: 10 tables in the classroom, with each table having 3 desktop servers running SCSI hardware and a pair of 10/100 network cards. At least 2 computers per table will contain a TDM40B card. Each student in the class also has a laptop which is connected to the LAN as well, so each table now has 3 desktops and 3 laptops. Each table then has network drops to both Internet and to LAN to racks at the back of the classroom. The racks contain approximately 20 Cisco 2600 routers and about 12 Cisco 3500 XL switches with fiber modules. There is also a single 6509 which we were using for Inter-VLAN routing over fiber between the switches. So basically, if you had access to this equipment, and you were asked to design some labs, what would you do? What kinds of things would you do with Asterisk based on what I have mentioned? If I have no given enough information somewhere, feel free to ask and I will explain anything that I can. Thanks in advance, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' onRedHat 9
On Tue, 2003-09-02 at 22:50, [EMAIL PROTECTED] wrote: > On Tue, Sep 02, 2003 at 09:35:34PM -0600, Gavin Hollinger wrote: > > >> > Sounds like an IRQ issue. > > > Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but > > > > Looks ok? > > > > > > cat /proc/interrupts > >CPU0 CPU1 > > 0: 9228581476395IO-APIC-edge timer > > 1: 0 4IO-APIC-edge keyboard > > 2: 0 0 XT-PIC cascade > > 4: 6 2IO-APIC-edge serial > > 8: 1 0IO-APIC-edge rtc > > 14: 7018 8073IO-APIC-edge ide0 > > 15: 33 3IO-APIC-edge ide1 > > 16: 11977985 11986942 IO-APIC-level t4xxp > > Looks a bit to high, but that might be standard. how quickly does it rise if you > do watch /proc/interrupts? if it raises rather quickly, I'd say its an irq > problem. change the location of the boards if possible, and if you don't need > things like the serial port, disable those in the bios. Combination of those > should get the problem resolved. Zapata cards make 8000 irqs a second. There is no buffer so after each cycle, the computer must service the card. This is true for x100p and s100U and the others as well as the digital cards. I do believe these counters roll over also. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
It only core's when u use the gatekeeper component, due to the way pwlib deals with memory allocation. This is going to take quite a lot of trying various different incantations to fix, unfortunately I cannot justify dedicating that kind time, at this point. Sorry, Jeremy McNamara Martin Pycko wrote: This happens only on relaod. You can disable reload routine in chan_h323.c ... Martin On 1 Sep 2003, Michael wrote: I'm running the CVS from last week and from day one (over 4 months now) I've had this problem where asterisk core dumps when using chan_h323. It appears to be a problem with pwlib and the console, but I'm not sure how to read the below output from gdb. I can start Asterisk just fine and chan_h323 works great when sending and receiving calls. I only have this core dump problem when sending a reload to Asterisk via the CLI or "asterisk -rx "reload"". Environment paths: LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 Core dump info: (gdb) bt #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PIntArray const&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #1 0x48315a2a in PSocket::Select(PSocket::SelectList&, PSocket::SelectList&, PSocket::SelectList&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x483151a7 in PSocket::Select(PSocket::SelectList&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper&, H323RasPDU&, H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const&, H323Transport*) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41efe680 "65.39.220.195", secret=0x41efe700 "") at ast_h323.cpp:949 #8 0x41eeed81 in reload () at chan_h323.c:1595 #9 0x08055362 in ast_module_reload () at loader.c:159 #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at cli.c:105 #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
On Tue, Sep 02, 2003 at 09:35:34PM -0600, Gavin Hollinger wrote: > >> > Sounds like an IRQ issue. > > Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but > > Looks ok? > > > cat /proc/interrupts >CPU0 CPU1 > 0: 9228581476395IO-APIC-edge timer > 1: 0 4IO-APIC-edge keyboard > 2: 0 0 XT-PIC cascade > 4: 6 2IO-APIC-edge serial > 8: 1 0IO-APIC-edge rtc > 14: 7018 8073IO-APIC-edge ide0 > 15: 33 3IO-APIC-edge ide1 > 16: 11977985 11986942 IO-APIC-level t4xxp Looks a bit to high, but that might be standard. how quickly does it rise if you do watch /proc/interrupts? if it raises rather quickly, I'd say its an irq problem. change the location of the boards if possible, and if you don't need things like the serial port, disable those in the bios. Combination of those should get the problem resolved. > 19: 8708 8683 IO-APIC-level usb-ohci, eth0 > NMI: 0 0 > LOC:23990312399029 > ERR: 0 > MIS: 0 > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
>> > Sounds like an IRQ issue. > Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but Looks ok? cat /proc/interrupts CPU0 CPU1 0: 9228581476395IO-APIC-edge timer 1: 0 4IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 4: 6 2IO-APIC-edge serial 8: 1 0IO-APIC-edge rtc 14: 7018 8073IO-APIC-edge ide0 15: 33 3IO-APIC-edge ide1 16: 11977985 11986942 IO-APIC-level t4xxp 19: 8708 8683 IO-APIC-level usb-ohci, eth0 NMI: 0 0 LOC:23990312399029 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
On Tue, Sep 02, 2003 at 10:10:17PM -0500, Peter Pauly wrote:
> On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
> > > correctly from X-lite but nothing else happens - no audio is
> > > heard. My understanding is that I should hear some sort of
> >
> > I am using x-lite with the asterisk demo no problem. All I modified was
> > sip.conf
> >
> > Is the asterisk server and your x-lite client on the same LAN segment?
> >
> > Is all iptables and firewall code turned off on the asterisk server?
> >
> > Gavin Hollinger
> >
> Here is the message I am getting from Asterisk:
>
> *CLI> -- Executing VoiceMail("SIP/2000-3296", "u1234") in new stack
> == Parsing '/etc/asterisk/voicemail.conf': Found
> -- Playing 'vm/1234/unavail'
> -- Playing 'vm-intro'
> -- Playing 'beep'
> -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0001
> WARNING[229391]: File app_voicemail.c, Line 673 (leave_voicemail): No audio
> available on SIP/2000-3296??
> -- User hung up
>
> It shows it is playing the files, but nothing is heard on the Xlite SIP software
> side.
>
Hmm. this rings a bell, try putting nat=yes in your sip.conf, I think that fixed
the problem for me. (Or was the the login timed out thing? *shrug*)
> When Asterisk starts up, it complains about OSS and ALSA problems -
> sound capabilities on the console are irrelavent in this case aren't
> they?
>
> I've tried deactivating several different codecs in X-lite - it doesn't help.
>
> Peter.
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Re: [Asterisk-Users] Still no audio on SIP phone
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
> > correctly from X-lite but nothing else happens - no audio is
> > heard. My understanding is that I should hear some sort of
>
> I am using x-lite with the asterisk demo no problem. All I modified was
> sip.conf
>
> Is the asterisk server and your x-lite client on the same LAN segment?
>
> Is all iptables and firewall code turned off on the asterisk server?
>
> Gavin Hollinger
>
Here is the message I am getting from Asterisk:
*CLI> -- Executing VoiceMail("SIP/2000-3296", "u1234") in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/1234/unavail'
-- Playing 'vm-intro'
-- Playing 'beep'
-- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0001
WARNING[229391]: File app_voicemail.c, Line 673 (leave_voicemail): No audio available
on SIP/2000-3296??
-- User hung up
It shows it is playing the files, but nothing is heard on the Xlite SIP software side.
When Asterisk starts up, it complains about OSS and ALSA problems -
sound capabilities on the console are irrelavent in this case aren't
they?
I've tried deactivating several different codecs in X-lite - it doesn't help.
Peter.
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Re: [Asterisk-Users] Sip Software from Nero Folk?
That's OK... we all start somewhere. And frankly, I'm not that far behind you. It is interesting to see so many people who are new to Asterisk join the list: I think it suggests a good future for Asterisk. Tim On Tuesday 02 September 2003 08:36 pm, Frank Latini wrote: > Thanks..that was the question...too new to understand. > - Original Message - > From: "Timothy Soos" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, September 02, 2003 22:27 > Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > > On Tuesday 02 September 2003 08:19 pm, Frank Latini wrote: > > > Hi all, > > > > > > New to the list. We are going to begin testing various voip gateways. > > I > > > > am trying to understand the reference to * in this thread. Is there a > > rule > > > > of the list that I need to be aware of ? Do not want to breech the > > > etiquette of the list. > > > > I am not totally sure I understand your question correctly, yet this > > answer > > > may help: > > > > The PBX system discussed here is called "Asterisk", and it is definately > > capable of doing VoIP. People often use the star symbol "*" as a > > short-hand > > > way of referring to the PBX system. > > -- > > Thanks, > > Timothy Soos > > XQL, LLC > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Timothy Soos XQL, LLC 303-480-8228 720-979-3128 (Direct) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Thanks..that was the question...too new to understand. - Original Message - From: "Timothy Soos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 22:27 Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > On Tuesday 02 September 2003 08:19 pm, Frank Latini wrote: > > Hi all, > > > > New to the list. We are going to begin testing various voip gateways. I > > am trying to understand the reference to * in this thread. Is there a rule > > of the list that I need to be aware of ? Do not want to breech the > > etiquette of the list. > > I am not totally sure I understand your question correctly, yet this answer > may help: > > The PBX system discussed here is called "Asterisk", and it is definately > capable of doing VoIP. People often use the star symbol "*" as a short-hand > way of referring to the PBX system. > -- > Thanks, > Timothy Soos > XQL, LLC > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Hi again, after reading more messages in the list. I get it! Thanks for the non-flames, etc. Frank - Original Message - From: "Frank Latini" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 22:19 Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > Hi all, > > New to the list. We are going to begin testing various voip gateways. I am > trying to understand the reference to * in this thread. Is there a rule of > the list that I need to be aware of ? Do not want to breech the etiquette > of the list. > > Thanks > > Frank > - Original Message - > From: "Gavin Hollinger" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, September 02, 2003 02:58 > Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > > > > haven't been able to get it to pass dtmf to *. I don't know if this > > > > Do you have > > dtmfmode=inband > > in sip.conf? > > > > http://www.sippstar.com/en/631927444894185.html > > > > Q.: DTMF generated by SIPPS is not recognized by other > > applications. > > > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > > application awaiting an event instead of a tone will not be able to work > > with SIPPS > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Software from Nero Folk?
* = Asterisk symbol = Asterisk = reference to Asterisk PBX -Original Message- From: Frank Latini [mailto:[EMAIL PROTECTED] Sent: Tue 9/2/2003 10:19 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? Hi all, New to the list. We are going to begin testing various voip gateways. I am trying to understand the reference to * in this thread. Is there a rule of the list that I need to be aware of ? Do not want to breech the etiquette of the list. Thanks Frank - Original Message - From: "Gavin Hollinger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 02:58 Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > haven't been able to get it to pass dtmf to *. I don't know if this > > Do you have > dtmfmode=inband > in sip.conf? > > http://www.sippstar.com/en/631927444894185.html > > Q.: DTMF generated by SIPPS is not recognized by other > applications. > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > application awaiting an event instead of a tone will not be able to work > with SIPPS > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users <>
Re: [Asterisk-Users] Sip Software from Nero Folk?
On Tuesday 02 September 2003 08:19 pm, Frank Latini wrote: > Hi all, > > New to the list. We are going to begin testing various voip gateways. I > am trying to understand the reference to * in this thread. Is there a rule > of the list that I need to be aware of ? Do not want to breech the > etiquette of the list. I am not totally sure I understand your question correctly, yet this answer may help: The PBX system discussed here is called "Asterisk", and it is definately capable of doing VoIP. People often use the star symbol "*" as a short-hand way of referring to the PBX system. -- Thanks, Timothy Soos XQL, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Hi all, New to the list. We are going to begin testing various voip gateways. I am trying to understand the reference to * in this thread. Is there a rule of the list that I need to be aware of ? Do not want to breech the etiquette of the list. Thanks Frank - Original Message - From: "Gavin Hollinger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 02:58 Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > haven't been able to get it to pass dtmf to *. I don't know if this > > Do you have > dtmfmode=inband > in sip.conf? > > http://www.sippstar.com/en/631927444894185.html > > Q.: DTMF generated by SIPPS is not recognized by other > applications. > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > application awaiting an event instead of a tone will not be able to work > with SIPPS > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
On Tue, Sep 02, 2003 at 03:37:03PM -0600, Gavin Hollinger wrote: > > Sounds like an IRQ issue. > > Perhaps. If that was the case, I should see an error somewhere right? > > Where could I look? > > Gavin Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but not for sound devices), its most likely a problem. try switching cards around in the box. -andrewg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Tones During Call
Hi, I am receiving calls via a Netjet-S card on asterisk, and I notice that whenever I am talkimng to someone, if their voice is loud enough, sometimes asterisk generates a DTMF Tone as they speak. that is played to me. (Caller doesn't hear it). Any ideas how to stop this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call answer confirmation
Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in zapata.conf -> Nothing happens. No ring is detected, the asnwer is not detected, only the hangup is detected and the call fails. Using "Answer confirmation" by entering "Channel: Zap/1c/XXX" in sample.call -> The # sign is correctly detected ans the answer is confirmed, but, it appears as if all other # sing used in the call are ignored. For instance, Comedian asks for the mailbox followed by the # sign, but the mailbox number is never acknowledge. Does anyone have a good experience with callprogress? Should I use some other hardware? Thank you.
[Asterisk-Users] ISDN
Hi,
I am using a Netjet-s ISDN Card, and am having some trouble dialling out
(Incoming Works Fine).
TRUNK=Modem/ttyI0
exten => _90X,1,Dial(${TRUNK}/${EXTEN:1}||Ttm)
exten => _90X,2,Congestion
I get the following when diallingout:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "Modem/ttyI0/04||Ttm") in new
stack
== Everyone is busy at this time
-- Executing Congestion("Zap/2-1", "") in new stack
== Spawn extension (local, 90422456118, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
I have tried inserting a "v" infront of the number, but to no avail.
Any Ideas??
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Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
On Tue, 2003-09-02 at 09:24, Martin Pycko wrote: > This happens only on relaod. You can disable reload routine in chan_h323.c > ... Thanks. I'll give it a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone compatible with Asterisk
On Monday 01 September 2003 02:18 pm, Tarun Banka wrote: > Hello All, > > I would like to know the most commonly used IP Phones with > Asterisk PBX. Your experience will help me in taking a right > decision to buy IP phones. > > Does anyone has experience with Telstrat i2732 IP Telephone and > SipPhone IP phones. Are these compatible with the ASterisk ? > > Any kind of pointers will help me alot in my investigation. I asked the good sir Mr. Malcolm Davenport a very similar question and I found his reply quite enlightening: "Asterisk supports all Analog telephones as well as SIP,H.323, and MGCP phones. SIP, H.323, and MGCP are VoIP protocols; SIP being the most popular at this time. ADSI phones are analog telephones that have large displays and programmable buttons. Asterisk offers support for ADSI phones. For more information on ADSI phones, send an e-mail to Greg Vance ([EMAIL PROTECTED]). The cheapest of the SIP phones on the market is the Grandstream Budgetone. It lists for $75 for the single-port and $85 for the dual-port model. The next step is the Snom 200 that lists for between $250 and $300. The Cisco 7960, my personal favorite, lists for between $350 and $450. And, the Pingtel Xpressa lists for $600." to that I will add an excellent article refereed by the good sir Mr. John Schmerold: http://www.nwc.com/shared/printArticle.jhtml?article=/1416/1416f2full.html&pub=nwc Maybe these items will help you. Thanks, Timothy Soos XQL, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' onRedHat 9
On Tue, 2003-09-02 at 17:11, Gavin Hollinger wrote: > > Note that this adds anything much to your problem. But I wanted to note > > that not all systems have hard times with systems working. This system > > Thanks for taking the time to do that for me. We are building a bigger > system that will be under heavier load with lots of 56k data. However, > your info gives me confidence that if I work through my configuration > problems it will work. I was beginning to get discouraged because I have > been "Testing Asterisk" for 2 months now. 8-( I guess the short coming > is mainly me, I will keep at it till I get it. > > What kernel / distribution are you using? It is a debian "unstable" system with custom compiled 2.4.20 kernel. Don't get too discouraged, My home asterisk machine isn't all that stable. Of course it is the one that runs bleeding edge code. The one I listed earlier is my switch machine, it is the only one I manage that is attached to the PSTN. All others including our office pbx do VoIP to that machine. With this much traffic on the one PSTN gateway, it only gets upgraded if I think it will fix a problem. So far that hasn't been needed. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
> Note that this adds anything much to your problem. But I wanted to note > that not all systems have hard times with systems working. This system Thanks for taking the time to do that for me. We are building a bigger system that will be under heavier load with lots of 56k data. However, your info gives me confidence that if I work through my configuration problems it will work. I was beginning to get discouraged because I have been "Testing Asterisk" for 2 months now. 8-( I guess the short coming is mainly me, I will keep at it till I get it. What kernel / distribution are you using? Thanks Gavin Hollinger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Configure DID Numbers with T1 Line & T100p
Kd, DID is handled like an extension: exten => 901212,1,Dial(SIP/[EMAIL PROTECTED]) A quick google search of the list was all I needed to find this... "site:lists.digium.com DID routing" -wade -Original Message- From: Kekin Dand [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 5:30 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Configure DID Numbers with T1 Line & T100p Hello Everyone, I am new to asterisk and linux too, I managed to installed asterisk on redhat8 with the help of mailing list archives and Handbook guide. I configured 2 SIP phones (grandstream) and it is working fine internally. We have T1 Line coming in with block of 200 DID Numbers. I want to assign DID Numbers to each of SIP phones as an extensions and able to call any PSTN line. I am not able to find, how to configure DID Block as an extension, which will route all calls from T100p card and T1 line. Any ides how do I configure DID's, where do I have to put entries for. Thanks, Any help is appreciated. Regards, Kd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openbsd PF firewall ?
i run it on freebsd. it has worked flawlessly. i prefer pf config file syntax over any of the others: ipfw[12], ipfilter and the linux variants. the pflog+pftcpdump feature is handy for seeing what your filter is denying. the only thing i miss in using it is dummynet. On 2003.09.02 17:17:12 +, marrandy wrote: > Hello. > > Trying firewalls out. > > Anyone had any success with an Openbsd PF firewall ? > > Regards...Martin > -- > Good news. Ten weeks from Friday will be a pretty good day. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' onRedHat 9
On Tue, 2003-09-02 at 16:17, Gavin Hollinger wrote: > TE410P - intermittent one way audio forcing reboot multiple times per day. > > > but have thought of restarting asterisk at cron time. If someone's on > > the phone at that time they're only talking to themselves anyway. > > > Yeah, that would work, except I am also having intermittent problems with > the same error during the day also. I cannot nail down or duplicate a > cause though. Just testing asterisk with the digium TE410P, I find myself > rebooting several times a day. I have another machine that does not have > any digium hardware that only needs to be restarted once every few weeks. > > Haven't got much response here, should this be posted as a bug report? Note that this adds anything much to your problem. But I wanted to note that not all systems have hard times with systems working. This system is on a 1200 celeron supermicro computer with a T400P card in it. phone:/home/critch# uptime 16:37:10 up 62 days, 5:04, 1 user, load average: 0.06, 0.08, 0.02 phone:/home/critch# asterisk -r Asterisk CVS-04/26/03-20:38:26, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <[EMAIL PROTECTED]> = Connected to Asterisk CVS-04 currently running on phone (pid = 1213) phone*CLI> show uptime System uptime: 8 weeks, 6 days, 5 hours, 6 minutes, 55 seconds Last reload: 2 weeks, 1 day, 52 minutes, 31 seconds phone*CLI> This is with a max in of 15 channels at a time from the PSTN. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openbsd PF firewall ?
On Tue, 2 Sep 2003, Jon Pounder wrote: > At 05:17 PM 9/2/2003 -0400, you wrote: > >Hello. > >Trying firewalls out. > >Anyone had any success with an Openbsd PF firewall ? > > works for us, seems fairly simple to configure, and tamper resistant since > it can run in bridge mode with no externally visible ips, so it is > impossible for an attacker to gain access to the machine through its > external interface. If you are more familiar with linux, I've got a writeup on how to do the same thing at http://www.sjdjweis.com/linux/bridging/ 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
> Sounds like an IRQ issue. Perhaps. If that was the case, I should see an error somewhere right? Where could I look? Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configure DID Numbers with T1 Line & T100p
Hello Everyone, I am new to asterisk and linux too, I managed to installed asterisk on redhat8 with the help of mailing list archives and Handbook guide. I configured 2 SIP phones (grandstream) and it is working fine internally. We have T1 Line coming in with block of 200 DID Numbers. I want to assign DID Numbers to each of SIP phones as an extensions and able to call any PSTN line. I am not able to find, how to configure DID Block as an extension, which will route all calls from T100p card and T1 line. Any ides how do I configure DID's, where do I have to put entries for. Thanks, Any help is appreciated. Regards, Kd
Re: [Asterisk-Users] Still no audio on SIP phone
> correctly from X-lite but nothing else happens - no audio is > heard. My understanding is that I should hear some sort of I am using x-lite with the asterisk demo no problem. All I modified was sip.conf Is the asterisk server and your x-lite client on the same LAN segment? Is all iptables and firewall code turned off on the asterisk server? Gavin Hollinger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' onRedHat 9
Sounds like an IRQ issue. bkw On Tue, 2 Sep 2003, Gavin Hollinger wrote: > TE410P - intermittent one way audio forcing reboot multiple times per day. > > > but have thought of restarting asterisk at cron time. If someone's on > > the phone at that time they're only talking to themselves anyway. > > > Yeah, that would work, except I am also having intermittent problems with > the same error during the day also. I cannot nail down or duplicate a > cause though. Just testing asterisk with the digium TE410P, I find myself > rebooting several times a day. I have another machine that does not have > any digium hardware that only needs to be restarted once every few weeks. > > Haven't got much response here, should this be posted as a bug report? > > Gavin Hollinger > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Openbsd PF firewall ?
At 05:17 PM 9/2/2003 -0400, you wrote: Hello. Trying firewalls out. Anyone had any success with an Openbsd PF firewall ? works for us, seems fairly simple to configure, and tamper resistant since it can run in bridge mode with no externally visible ips, so it is impossible for an attacker to gain access to the machine through its external interface. Regards...Martin -- Good news. Ten weeks from Friday will be a pretty good day. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
try system,"/usr 01 on"
or system("/usr .. on")
Martin
On Tue, 2 Sep 2003, Josh Edwards wrote:
> Question below, here is the file in question
> exten => 9,1,system,/usr/local/bin/hetest 01 on
> exten => 9,2,system,/usr/local/bin/hetest 02 on
> exten => 9,3,system,/usr/local/bin/hetest 03 on
> exten => 9,4,system,/usr/local/bin/hetest 04 on
> exten => 9,5,system,/usr/local/bin/hetest 05 on
> exten => 9,6,system,/usr/local/bin/hetest 06 on
> exten => 9,7,system,/usr/local/bin/hetest 07 on
> exten => 9,8,system,/usr/local/bin/hetest 08 on
> exten => 9,9,system,/usr/local/bin/hetest 09 on
>
> When I dial 9 it runs the first item, then it exits and gives me a busy,
> why does it not go through all of the items then exit
>
>
> Josh
>
>
> Get MSN 8 and help protect your children with advanced parental controls.
> ___ Asterisk-Users mailing
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>
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Re: [Asterisk-Users] TE410P - one way audio, after 'rpm -qa' on RedHat 9
TE410P - intermittent one way audio forcing reboot multiple times per day. > but have thought of restarting asterisk at cron time. If someone's on > the phone at that time they're only talking to themselves anyway. Yeah, that would work, except I am also having intermittent problems with the same error during the day also. I cannot nail down or duplicate a cause though. Just testing asterisk with the digium TE410P, I find myself rebooting several times a day. I have another machine that does not have any digium hardware that only needs to be restarted once every few weeks. Haven't got much response here, should this be posted as a bug report? Gavin Hollinger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Openbsd PF firewall ?
Hello. Trying firewalls out. Anyone had any success with an Openbsd PF firewall ? Regards...Martin -- Good news. Ten weeks from Friday will be a pretty good day. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
On Tue, 2003-09-02 at 15:48, Josh Edwards wrote: > Nothing, it is a shell script that runs another program. do I need to > have it return something? Yes, all applications should return a result code after finishing. Usually 0 means success or no errors and anything else is some form of error indicator. So also to be checked out from asterisk... -= Info about application 'System' =- [Synopsis]: Execute a system command [Description]: System(command): Executes a command by using system(). Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. >From the bash man page, but could probably be in any shell... exit [n] Cause the shell to exit with a status of n. If n is omitted, the exit status is that of the last command executed. A trap on EXIT is executed before the shell terminates. >From this you should note that ideally you should exit 0. > >From: Steven Critchfield > >Reply-To: [EMAIL PROTECTED] > >To: [EMAIL PROTECTED] > >Subject: Re: [Asterisk-Users] extensions.conf issue > >Date: Tue, 02 Sep 2003 15:11:34 -0500 > > > >On Tue, 2003-09-02 at 14:54, Josh Edwards wrote: > > > Question below, here is the file in question > > > exten => 9,1,system,/usr/local/bin/hetest 01 on > > > exten => 9,2,system,/usr/local/bin/hetest 02 on > > > exten => 9,3,system,/usr/local/bin/hetest 03 on > > > exten => 9,4,system,/usr/local/bin/hetest 04 on > > > exten => 9,5,system,/usr/local/bin/hetest 05 on > > > exten => 9,6,system,/usr/local/bin/hetest 06 on > > > exten => 9,7,system,/usr/local/bin/hetest 07 on > > > exten => 9,8,system,/usr/local/bin/hetest 08 on > > > exten => 9,9,system,/usr/local/bin/hetest 09 on > > > > > > > > > When I dial 9 it runs the first item, then it exits and gives me a > > > busy, why does it not go through all of the items then exit > > > >what does hetest return? > >-- > >Steven Critchfield > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > __ > Get MSN 8and enjoy automatic e-mail virus protection. > ___ Asterisk-Users mailing > list [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
Nothing, it is a shell script that runs another program. do I need to have it return something? Josh >From: Steven Critchfield <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] extensions.conf issue >Date: Tue, 02 Sep 2003 15:11:34 -0500 > >On Tue, 2003-09-02 at 14:54, Josh Edwards wrote: > > Question below, here is the file in question > > exten => 9,1,system,/usr/local/bin/hetest 01 on > > exten => 9,2,system,/usr/local/bin/hetest 02 on > > exten => 9,3,system,/usr/local/bin/hetest 03 on > > exten => 9,4,system,/usr/local/bin/hetest 04 on > > exten => 9,5,system,/usr/local/bin/hetest 05 on > > exten => 9,6,system,/usr/local/bin/hetest 06 on > > exten => 9,7,system,/usr/local/bin/hetest 07 on > > exten => 9,8,system,/usr/local/bin/hetest 08 on > > exten => 9,9,system,/usr/local/bin/hetest 09 on > > > > > > When I dial 9 it runs the first item, then it exits and gives me a > > busy, why does it not go through all of the items then exit > >what does hetest return? >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users Get MSN 8 and enjoy automatic e-mail virus protection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still no audio on SIP phone
I have been using X-Lite on FWD without any troubles and recently became interested in trying asterisk. I am able to register from X-Lite and dial a number - I've tried dialing some of the sample numbers in the sample extentions.conf file, like 500 and 1234, they appear to dial correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some sort of message. I already found one problem - on my debian system - /usr/bin/mpg123 was a symbolic link pointing to mpg321. I've corrected that and installed mpg123/unstable and made sure it was the real deal (deleted the symbolic link, etc). I am still not getting any audio. My setup: Debian (from Knoppix - a mix of unstable, testing, stable), no hardware phone cards, one software SIP phone (X-lite). Everything is on a LAN (no firewall involved). Where should I begin to find this problem? I've tried starting asterisk with lots of verbose flags, but I don't see anything suspicious. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Phone 7905G
Has anyone had any success using a Cisco 7905G phone with Asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
On Tue, 2003-09-02 at 14:54, Josh Edwards wrote: > Question below, here is the file in question > exten => 9,1,system,/usr/local/bin/hetest 01 on > exten => 9,2,system,/usr/local/bin/hetest 02 on > exten => 9,3,system,/usr/local/bin/hetest 03 on > exten => 9,4,system,/usr/local/bin/hetest 04 on > exten => 9,5,system,/usr/local/bin/hetest 05 on > exten => 9,6,system,/usr/local/bin/hetest 06 on > exten => 9,7,system,/usr/local/bin/hetest 07 on > exten => 9,8,system,/usr/local/bin/hetest 08 on > exten => 9,9,system,/usr/local/bin/hetest 09 on > > > When I dial 9 it runs the first item, then it exits and gives me a > busy, why does it not go through all of the items then exit what does hetest return? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
Try System("/usr/local/bin/hetest 01 on")
bkw
On Tue, 2 Sep 2003, Josh Edwards wrote:
> Question below, here is the file in question
> exten => 9,1,system,/usr/local/bin/hetest 01 on
> exten => 9,2,system,/usr/local/bin/hetest 02 on
> exten => 9,3,system,/usr/local/bin/hetest 03 on
> exten => 9,4,system,/usr/local/bin/hetest 04 on
> exten => 9,5,system,/usr/local/bin/hetest 05 on
> exten => 9,6,system,/usr/local/bin/hetest 06 on
> exten => 9,7,system,/usr/local/bin/hetest 07 on
> exten => 9,8,system,/usr/local/bin/hetest 08 on
> exten => 9,9,system,/usr/local/bin/hetest 09 on
>
> When I dial 9 it runs the first item, then it exits and gives me a busy, why does it
> not go through all of the items
> then exit
>
>
> Josh
>
> __
> Get MSN 8 and help protect your children with advanced parental controls.
> ___ Asterisk-Users mailing list [EMAIL
> PROTECTED]
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>
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[Asterisk-Users] extensions.conf issue
Question below, here is the file in question exten => 9,1,system,/usr/local/bin/hetest 01 onexten => 9,2,system,/usr/local/bin/hetest 02 onexten => 9,3,system,/usr/local/bin/hetest 03 onexten => 9,4,system,/usr/local/bin/hetest 04 onexten => 9,5,system,/usr/local/bin/hetest 05 onexten => 9,6,system,/usr/local/bin/hetest 06 onexten => 9,7,system,/usr/local/bin/hetest 07 onexten => 9,8,system,/usr/local/bin/hetest 08 onexten => 9,9,system,/usr/local/bin/hetest 09 on When I dial 9 it runs the first item, then it exits and gives me a busy, why does it not go through all of the items then exit Josh Get MSN 8 and help protect your children with advanced parental controls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stuck On ISDN
Hi Olle, the cheapes CAPI card is the passive AVM Fritz Card PCI. It's nice to start playing, but for a production system (isdn pbx) i can only recommend the active Eicon Diva Server cards (which support EC). regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Die, 2003-09-02 um 21.36 schrieb Olle E. Johansson: > The devices that support CAPI seems much more expensive than ISDN Cards > with I4L support. What's the least expensive ISDN card with support for CAPI? > > I finally got I4L working and the sound quality could be better, but it > works for experimental systems. Since I do not _really_ understand how > I got it working from a state with no reaction at all I can't explain > or assist, just send my config file on request. > > The manual says nothing on the subject on I4L support and there's > nothing to find on the net by googling... Everyone points to > capi and, back to the start of my reply, it seems expensive for personal > use... > > /Olle > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] exiting Voicemai for VoiceMailMainl
Title: Message Has anyone been able to exit voicemail2 to get to voicemailmain2 in order to check voicemail messages from a remote phone? I can dial a "0" while my voicemail intro plays, and * dumps the call and says it was sent into invalid extension "o". I set up extension "0", and I can call it from any phone on the system. I can transfer any phone to it. The "0" extensionis set up to enter voicemailmain2. It works fine. Is there another way to exit during the voice mail prompt in order to check your voic mail? Thanks, Jerry
Re: [Asterisk-Users] Stuck On ISDN
Tomas Prybil wrote: max power wrote: Spent that last week or so trying to get isdn4linux working. how do I link ttyIO to asterisk?I cannot dial out or dialin. I can see the call coming in in /var/log/messages. Has anyone any tips? I am not familar with isdn4linux. What kind of IDDN device are you using? I would recommend to use CAPI instead of i4 if your device supports it. Works like a charm. Have a look at http://www.junghanns.net/asterisk/ for some inspiration :) The devices that support CAPI seems much more expensive than ISDN Cards with I4L support. What's the least expensive ISDN card with support for CAPI? I finally got I4L working and the sound quality could be better, but it works for experimental systems. Since I do not _really_ understand how I got it working from a state with no reaction at all I can't explain or assist, just send my config file on request. The manual says nothing on the subject on I4L support and there's nothing to find on the net by googling... Everyone points to capi and, back to the start of my reply, it seems expensive for personal use... /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stuck On ISDN
max power wrote: Spent that last week or so trying to get isdn4linux working. how do I link ttyIO to asterisk?I cannot dial out or dialin. I can see the call coming in in /var/log/messages. Has anyone any tips? I am not familar with isdn4linux. Hi. What kind of IDDN device are you using? I would recommend to use CAPI instead of i4 if your device supports it. Works like a charm. Have a look at http://www.junghanns.net/asterisk/ for some inspiration :) /t Max... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with mediatrix 1204 FXO
Try canreinvite=no
Martin
On Tue, 2 Sep 2003, Zac Sprackett wrote:
> I'm having a problem getting outbound trunking to work using asterisk
> and an external SIP FXO.
>
> 7 digit dialing produces the following output:
>
> -- Executing Dial("SIP/mitel-fe17", "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17
> -- Attempting native bridge of SIP/mitel-fe17 and SIP/mediatrix-1204-645e
> -- Got SIP response 481 "Call-Leg Does Not Exist" back from 172.20.16.7
> == Spawn extension (internal, 5925660, 1) exited non-zero on 'SIP/mitel-fe17'
>
> THe PSTN gateway is siezing the trunk and dialing the call. The native
> bridging seems to be the point of failure. The caller (another sip set)
> gets hung up on and the called pary hears dead air.
>
> A sip debug log of the scenario is available here:
>
> http://sprackett.com/asterisk.txt
>
> extensions.conf snippet:
>
> [trunks-local]
> ; local 7 digit dialing
> exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
> exten => _NXX,2,Congestion
>
> [internal]
> include => trunks-local
>
>
> sip.conf snippet:
>
> [mediatrix-1204]
> type=peer
> host=172.20.16.7
> mask=255.255.255.255
> dtmfmode=inband
> context=default
> canreinvite=yes
> qualify=yes
>
> Thanks in advance for any adive you can give me.
> -z
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>
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Re: [Asterisk-Users] Low bit rate codec (speex)
Well put.
On Tue, 2 Sep 2003, John Todd wrote:
> At 11:42 AM -0500 9/2/03, Brian West wrote:
> >
> >http://bugs.digium.com/bug_view_page.php?bug_id=149
> >
> >bkw
> >
> >On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
> >
> >> On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
> >> Brian West <[EMAIL PROTECTED]> wrote:
> >>
> >> > I opened a request on bugs.digium.com for this feature. The 6k and 8k
> >> > codecs are very impressive also.
> >> >
> > > > bkw
> > >
> >>Where can I see the status of this request?
> >>
> >> []'s
> > > Eduardo
>
> Spoiler: Read to the bottom about how to get 400 calls into a megabit. Maybe.
>
> I'm ashamed to say I had not actually looked at the lower-bandwidth
> encoding options of Speex in the past, and skipped right over that
> section of the text in favor of the high bandwidth bitrates. What a
> mistake! I am extremely impressed with the 4kbps VBR rate for Speex,
> at least from the samples on the Speex website.
>
> If the sound quality is as good as advertised at the low bitrates,
> the addition of selectable features for Speex would truly be an asset
> to Asterisk on a per-call basis (heck, even just per-peer.) I have
> several clients who need to move traffic across international IP
> capacity, and the low-bandwidth option of choice to them is G.729
> (LPC10 is not an option due to sound quality issues.) The very
> interesting features of VAD and VBR look to be (on paper, at least) a
> real win as well, with the channel bitrate being reduced even further
> by silence and sound complexity compression.
>
> Exposing codec feature selections to the dialplan would be
> interesting, but I expect Mark will want to (correctly) implement a
> generic method for doing this. However, are there any other codecs
> that Asterisk supports that have the ability to use different options
> (bitrate, VBR, VAD)? Is it worthwhile to make this a generic
> function of some sort, or is it sufficient to make specific
> techniques just for Speex? (${SPEEX-BITRATE}, or ${SPEEX-VBR} to give
> crude examples.)
>
> Why am I so excited about this? The point of VoIP for most of my
> customers is twofold: the first point is the addition of new and
> novel services that they would not be able to offer previously
> without investing a lot in hardware. The second (and for some, the
> primary) point is that VoIP allows the transmission of voice packets
> over a less expensive packetized path than TDM. Thus, the biggest
> number on their minds is "Cost of Bandwidth!" The more voice
> streams you can pack into the bandwidth, the less they pay for
> bandwidth, and thus the larger the profit margin - very simple
> equation. So, they're really REALLY interested in any way to get
> more calls into the same number of bits per second, and Speex seems
> to have some interesting options in that arena.
>
> Combined with the clever use of trunking with IAX2, I could possibly
> see (looking at back-of-napkin, totally theoretical numbers)
> something like 400 calls in a megabit between two Asterisk servers.
> That number seems wrong to me, and I expect my first impression is
> correct, but here's the math: with my IAX2 tests which I documented
> previously on this list, I got a theoretical 103 calls into a megabit
> of bandwidth with G.729 at 9.6kbps per additional call. Now, the
> Speex codec can be turned down to 4kbps, so I can get 2.4 Speex calls
> into the same space that I fit one G.729 call. So, (2.4 * 103 = 247)
> into a megabit. Now, usually only one person is talking at a time.
> This means VAD would be active on 50% of the channels, thus
> eliminating traffic in one way for all calls. I'm sure that's not
> quite accurate due to background noise and overtalk, so let's say
> that only 30% of the legs are empty at any one time due to VAD. So,
> that's an additional few channels, so now we're at (247 + (247 * .3)
> = 321) total channels. Now I move into the really unstable math
> (i.e.: I'm making this up based on wild fantasy.) If VBR is
> implemented, maybe/hopefully/possibly that permits us another 25%
> savings on bits per second, so that turns into (321 + (321 * .25) =
> 401) channels in a single megabit. This seems impossible. Anyone
> care to shoot holes in these numbers?
>
> JT
>
>
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>
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Re: [Asterisk-Users] Low bit rate codec (speex)
At 11:42 AM -0500 9/2/03, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=149
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
Brian West <[EMAIL PROTECTED]> wrote:
> I opened a request on bugs.digium.com for this feature. The 6k and 8k
> codecs are very impressive also.
>
> > bkw
>
Where can I see the status of this request?
[]'s
> Eduardo
Spoiler: Read to the bottom about how to get 400 calls into a megabit. Maybe.
I'm ashamed to say I had not actually looked at the lower-bandwidth
encoding options of Speex in the past, and skipped right over that
section of the text in favor of the high bandwidth bitrates. What a
mistake! I am extremely impressed with the 4kbps VBR rate for Speex,
at least from the samples on the Speex website.
If the sound quality is as good as advertised at the low bitrates,
the addition of selectable features for Speex would truly be an asset
to Asterisk on a per-call basis (heck, even just per-peer.) I have
several clients who need to move traffic across international IP
capacity, and the low-bandwidth option of choice to them is G.729
(LPC10 is not an option due to sound quality issues.) The very
interesting features of VAD and VBR look to be (on paper, at least) a
real win as well, with the channel bitrate being reduced even further
by silence and sound complexity compression.
Exposing codec feature selections to the dialplan would be
interesting, but I expect Mark will want to (correctly) implement a
generic method for doing this. However, are there any other codecs
that Asterisk supports that have the ability to use different options
(bitrate, VBR, VAD)? Is it worthwhile to make this a generic
function of some sort, or is it sufficient to make specific
techniques just for Speex? (${SPEEX-BITRATE}, or ${SPEEX-VBR} to give
crude examples.)
Why am I so excited about this? The point of VoIP for most of my
customers is twofold: the first point is the addition of new and
novel services that they would not be able to offer previously
without investing a lot in hardware. The second (and for some, the
primary) point is that VoIP allows the transmission of voice packets
over a less expensive packetized path than TDM. Thus, the biggest
number on their minds is "Cost of Bandwidth!" The more voice
streams you can pack into the bandwidth, the less they pay for
bandwidth, and thus the larger the profit margin - very simple
equation. So, they're really REALLY interested in any way to get
more calls into the same number of bits per second, and Speex seems
to have some interesting options in that arena.
Combined with the clever use of trunking with IAX2, I could possibly
see (looking at back-of-napkin, totally theoretical numbers)
something like 400 calls in a megabit between two Asterisk servers.
That number seems wrong to me, and I expect my first impression is
correct, but here's the math: with my IAX2 tests which I documented
previously on this list, I got a theoretical 103 calls into a megabit
of bandwidth with G.729 at 9.6kbps per additional call. Now, the
Speex codec can be turned down to 4kbps, so I can get 2.4 Speex calls
into the same space that I fit one G.729 call. So, (2.4 * 103 = 247)
into a megabit. Now, usually only one person is talking at a time.
This means VAD would be active on 50% of the channels, thus
eliminating traffic in one way for all calls. I'm sure that's not
quite accurate due to background noise and overtalk, so let's say
that only 30% of the legs are empty at any one time due to VAD. So,
that's an additional few channels, so now we're at (247 + (247 * .3)
= 321) total channels. Now I move into the really unstable math
(i.e.: I'm making this up based on wild fantasy.) If VBR is
implemented, maybe/hopefully/possibly that permits us another 25%
savings on bits per second, so that turns into (321 + (321 * .25) =
401) channels in a single megabit. This seems impossible. Anyone
care to shoot holes in these numbers?
JT
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[Asterisk-Users] problems with mediatrix 1204 FXO
I'm having a problem getting outbound trunking to work using asterisk
and an external SIP FXO.
7 digit dialing produces the following output:
-- Executing Dial("SIP/mitel-fe17", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/mediatrix-1204-645e answered SIP/mitel-fe17
-- Attempting native bridge of SIP/mitel-fe17 and SIP/mediatrix-1204-645e
-- Got SIP response 481 "Call-Leg Does Not Exist" back from 172.20.16.7
== Spawn extension (internal, 5925660, 1) exited non-zero on 'SIP/mitel-fe17'
THe PSTN gateway is siezing the trunk and dialing the call. The native
bridging seems to be the point of failure. The caller (another sip set)
gets hung up on and the called pary hears dead air.
A sip debug log of the scenario is available here:
http://sprackett.com/asterisk.txt
extensions.conf snippet:
[trunks-local]
; local 7 digit dialing
exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => _NXX,2,Congestion
[internal]
include => trunks-local
sip.conf snippet:
[mediatrix-1204]
type=peer
host=172.20.16.7
mask=255.255.255.255
dtmfmode=inband
context=default
canreinvite=yes
qualify=yes
Thanks in advance for any adive you can give me.
-z
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RE: [Asterisk-Users] Installation Problem
On Tue, 2003-09-02 at 12:07, John Todd wrote: > >On Tue, 2003-09-02 at 02:27, John Todd wrote: > >> Phil - > >> Here are my "generic" notes and reminders for Asterisk on Debian. > >> These may be hacks; your mileage may vary. > >> > >> debian asterisk install notes: > >> - in asterisk/Makefile: added "-I/usr/local/ssl/include" to CFLAGS line > >> - in asterisk/res/Makefile: added "-L/usr/local/ssl/lib" to CRYPTO_LIBS line > >> - in zaptel/Makefile: commented out KFLAGS+=-DCONFIG_ZAPATA_PPP line > >> - installed libnewt-dev > >> - installed newt-tcl (?needed) > >> - installed "apt-get source openssl" > >> - installed "apt-get install openssl" > > > >There are dev packages for openssl so the the source is not necessary. > >The dev packages put the headers in the right place and therefore the > >/usr/local/ changes aren't necessary. > > > >libnewt-dev is important so you can compile zttool and astman. newt-tcl > >shouldn't be necessary, but may be a debian dependency thing. > > > >The PPP line is only needed if either your kernel doesn't have PPP > >support, or if you don't plan on using the PPP options on the T1/E1 > >interfaces. > > > >Hope that clears up any problems. > >-- > >Steven Critchfield <[EMAIL PROTECTED]> > > OK, thanks for the follow-up. I just did what was required to get it > running on the particular version of Debian that I was given. I am > unfamiliar with the distro, so those are my notes that I used to get > it working. For whatever reason, the ssl libraries were not found > correctly, and I had to modify the Makefiles to do the right thing. > The version I was using didn't have the ppp support in the kernel by > "default", so I deactivated it - none of my clients use the RAS > features of Asterisk, so it's no big loss. > > The one last thing I did notice is that after someone else has > installed "new" kernels, the "/usr/src/linux" symlink to the kernel > directory in the same . went away. I don't know if this is part of a > normal kernel upgrade with Debian or what, but I've had to link it > manually twice to get Asterisk to compile. I wonder if it wouldn't be better to look for the kernel headers in the /lib/modules/2.x.x/build/include directory. This is the way VMWare does it's default build against the currently running kernel. The current way the linus tree is released is to untar to a versioned directory where you then are able to make a unversioned/latest link to the one you are using. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message-waiting-indicator thru ZAPinterfaces - how to?
On Tue, 2 Sep 2003, John Todd wrote: >I'm trying to make the MWI indicators on my client's Vodavi Starplus >DHS phones work. The actual signalling - in-band DTMF from the ZAP >interfaces directly to the PBX system - works fine. I can manually >tell asterisk to send "#9610" as DTMF and voila, the MWI on >extension 10 lights (or goes out.) The question is, how is this >integrated with voicemail, i.e. so that the MWI turns on and off >appropriately, when new messages arrive and after a user has >listened to their messages? > >I've checked the last two months of mailing-list messages but found >no mention of this situation. Any tips or pointers to online docs >would be appreciated. > >Thanks, > >Sam > >P.S. Thanks to Jsmith for the fast, simple answer to my last >question re: version number in CVS not updating. I'm afraid that the answer to this, without programming some stuff inside of Asterisk, is uuugly. I suspect it will involve using perl or shell scripts to actually peek inside of the /var/spool/asterisk/vm directories and check things manually, out of a cron job or out of the "h" context with a System call. Then, a call would be created by the script (see sample.call) - just thinking about this method gives me the willies. The clean way would be to put a tiny call into the voicemail app (or would it be in app.c?) that triggers an outbound call with the appropriate parameters (see sample.call) I was working with someone over the weekend that is working on something like this, and it might even be the same type of system because the DTMF trigger looks similar. The easiest way to do this is with an external daemon that connects to the manager interface. You just need to watch for MessageWaiting events and when you see a change of state trigger an Originate action to dial out and enter the required DTMF. James Yes, that would work just as well. However, you'd also still need to write a "harvester" that would check the status of each extension. You can never trust that your runtime daemon has "caught" all the messages that may have been created, and then there's also the problem of handling messages that are already in the system when your daemon is (re)launched. Probably a combination: the daemon that listens to the manager events, and then every XX hours (and on startup) the harvester would check all possible mailboxes for statuses to ensure that things were "correct". This is a typical configuration for stateful checks on semi-realtime systems like this. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stuck On ISDN
Spent that last week or so trying to get isdn4linux working. how do I link ttyIO to asterisk?I cannot dial out or dialin. I can see the call coming in in /var/log/messages. Has anyone any tips? I am not familar with isdn4linux. Max... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installation Problem
On Tue, 2003-09-02 at 02:27, John Todd wrote: Phil - Here are my "generic" notes and reminders for Asterisk on Debian. These may be hacks; your mileage may vary. debian asterisk install notes: - in asterisk/Makefile: added "-I/usr/local/ssl/include" to CFLAGS line - in asterisk/res/Makefile: added "-L/usr/local/ssl/lib" to CRYPTO_LIBS line - in zaptel/Makefile: commented out KFLAGS+=-DCONFIG_ZAPATA_PPP line - installed libnewt-dev - installed newt-tcl (?needed) - installed "apt-get source openssl" - installed "apt-get install openssl" There are dev packages for openssl so the the source is not necessary. The dev packages put the headers in the right place and therefore the /usr/local/ changes aren't necessary. libnewt-dev is important so you can compile zttool and astman. newt-tcl shouldn't be necessary, but may be a debian dependency thing. The PPP line is only needed if either your kernel doesn't have PPP support, or if you don't plan on using the PPP options on the T1/E1 interfaces. Hope that clears up any problems. -- Steven Critchfield <[EMAIL PROTECTED]> OK, thanks for the follow-up. I just did what was required to get it running on the particular version of Debian that I was given. I am unfamiliar with the distro, so those are my notes that I used to get it working. For whatever reason, the ssl libraries were not found correctly, and I had to modify the Makefiles to do the right thing. The version I was using didn't have the ppp support in the kernel by "default", so I deactivated it - none of my clients use the RAS features of Asterisk, so it's no big loss. The one last thing I did notice is that after someone else has installed "new" kernels, the "/usr/src/linux" symlink to the kernel directory in the same . went away. I don't know if this is part of a normal kernel upgrade with Debian or what, but I've had to link it manually twice to get Asterisk to compile. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unified Messaging Support ?
My point as that you would need to pass the fax traffic through via the f extension to nother port which would be back to back connected to something that is a faxmodem and talks to hylafax Another option are the Eicon DIVA Server cards which could do the needed stuff via CAPI. On Tuesday 02 September 2003 6:33 pm, Wade J. Weppler wrote: > HylaFAX needs to connect to a modem, and the modem in turn needs to > connect to a phoneline. This phoneline has to be "real" so you can't > use a dummy driver. The TDM400P or T100P/E100P/T400P/TE410P+channel > bank would be the only suitable (and supported) choices for analog modem > connections. > > -wade > > > -Original Message- > > From: Lee Goodman [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, September 02, 2003 12:24 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Unified Messaging Support ? > > > > Do the ports have to be real (like XP100) or can you use the ztdummy > > (the > > > dummy zapatel driver) for the ports required for hylafax? > > > > Lee Goodman > > - Original Message - > > From: "Michael Bielicki" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, September 02, 2003 4:18 AM > > Subject: Re: [Asterisk-Users] Unified Messaging Support ? > > > > > On Monday 01 September 2003 4:28 am, [EMAIL PROTECTED] wrote: > > > > On Tue, 1 Sep 2003, Tarun Banka wrote: > > > > > One quick question. Does anyone has experience implementing > > > > > unified messaging (UM) using Asterisk. Does Asterisk has support > > > > > for UM ? > > > > > > > > Not really. Half of it works. If you mean by UM (Voicemail > > integrated > > > to > > > > > > your email box, and then send an SMS/Page etc to you) then its all > > > > doable. > > > > > > The problem happens when you want to bring fax into the picture. > > That > > > > > cannot be handled currently, as in, * can't take a fax, and route > > it > > > to > > > > > > your email as a PDF. > > > > > > > > But, you could use a third-party fax thingamajig and I'm sure > > connect > > > it > > > > > > to * for a good UM solution. > > > > > > Just pass it to hylafax and you fly, but it requires some planning > > cause > > > you > > > > > will need ddouble the amount of ports plus the fax devices for > > hylafax > > > > > -- > > > > wasim > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low bit rate codec (speex)
http://bugs.digium.com/bug_view_page.php?bug_id=149 bkw On Tue, 2 Sep 2003, Eduardo Goncalves wrote: > On Tue, 2 Sep 2003 10:27:53 -0500 (CDT) > Brian West <[EMAIL PROTECTED]> wrote: > > > I opened a request on bugs.digium.com for this feature. The 6k and 8k > > codecs are very impressive also. > > > > bkw > > > > Where can I see the status of this request? > > []'s > Eduardo > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unified Messaging Support ?
HylaFAX needs to connect to a modem, and the modem in turn needs to connect to a phoneline. This phoneline has to be "real" so you can't use a dummy driver. The TDM400P or T100P/E100P/T400P/TE410P+channel bank would be the only suitable (and supported) choices for analog modem connections. -wade > -Original Message- > From: Lee Goodman [mailto:[EMAIL PROTECTED] > Sent: Tuesday, September 02, 2003 12:24 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Unified Messaging Support ? > > Do the ports have to be real (like XP100) or can you use the ztdummy (the > dummy zapatel driver) for the ports required for hylafax? > > Lee Goodman > - Original Message - > From: "Michael Bielicki" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, September 02, 2003 4:18 AM > Subject: Re: [Asterisk-Users] Unified Messaging Support ? > > > > On Monday 01 September 2003 4:28 am, [EMAIL PROTECTED] wrote: > > > On Tue, 1 Sep 2003, Tarun Banka wrote: > > > > One quick question. Does anyone has experience implementing > > > > unified messaging (UM) using Asterisk. Does Asterisk has support > > > > for UM ? > > > > > > Not really. Half of it works. If you mean by UM (Voicemail integrated > to > > > your email box, and then send an SMS/Page etc to you) then its all > doable. > > > > > > The problem happens when you want to bring fax into the picture. That > > > cannot be handled currently, as in, * can't take a fax, and route it > to > > > your email as a PDF. > > > > > > But, you could use a third-party fax thingamajig and I'm sure connect > it > > > to * for a good UM solution. > > Just pass it to hylafax and you fly, but it requires some planning cause > you > > will need ddouble the amount of ports plus the fax devices for hylafax > > > > > > -- > > > wasim > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Packet8 DTA310
Well this debug desn't show the bad call setup. And furthermore all commands are accepted by the asterisk/UA. Martin On Mon, 1 Sep 2003, Andrew Joakimsen wrote: > There might be some other stuff mixed in there as well, 64.36.104.205 is > asterisk and 64.36.104.206 is the DTA > > 11 headers, 2 lines > Reliably Transmitting: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 > From: "asterisk" ;tag=as17328ab1 > To: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 36 > > Messages-Waiting: no > Voicemail: 0/1 > (no NAT) to 64.36.104.203:5060 > Sip read: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161 > From: "asterisk" ;tag=as17328ab1 > To: ;tag=6b0caf74-4936-bceb-18dc-3ee5469aa3f6 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Grandstream SIP UA 1.0.3.81 > Contact: > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE > Content-Length: 0 > > > 10 headers, 0 lines > Sip read: > SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport > From: ;tag=t2d9e0a11a85c88g > To: sip:[EMAIL PROTECTED] > Call-ID: [EMAIL PROTECTED] > CSeq: 100 SUBSCRIBE > Contact: sip:[EMAIL PROTECTED] > Expires: 3600 > Max-Forwards: 70 > Event: traverse > User-Agent: DTA SIP/0.11.8 NNOS/VR30 > Content-Type: application/sdp > Content-Length: 156 > > v=0 > o=0403532579 0 0 IN IP4 64.36.104.206 > =-m3*CLI> > c=IN IP4 64.36.104.206 > t=0 0 > m=audio 8002 RTP/AVP 18 101 > a=ptime:10 > a=rtpmap:101 telephone-event/8000 > > 13 headers, 8 lines > Using latest SUBSCRIBE request as basis request > Sending to 64.36.104.206 : 5060 (non-NAT) > Looking for in international > Transmitting (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.206;branch=z9hG4bK.2ddac.11a85c89;rport > From: ;tag=t2d9e0a11a85c88g > To: sip:[EMAIL PROTECTED];tag=as57545bcd > Call-ID: [EMAIL PROTECTED] > CSeq: 100 SUBSCRIBE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 3600 > Contact: ;expires=3600 > Content-Length: 0 > > > to 64.36.104.206:5060 > Reliably Transmitting: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 > From: sip:[EMAIL PROTECTED];tag=as57545bcd > To: ;tag=t2d9e0a11a85c88g > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Content-Type: application/xpidf+xml > Content-Length: 352 > > > "xpidf.dtd"> > > > > > > > > > > (no NAT) to 64.36.104.206:5060 > Sip read: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK5bb88bc0 > From: ;tag=as57545bcd > To: ;tag=t2d9e0a11a85c88g > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > Server: DTA SIP/0.11.8 NNOS/VR30 > Content-Length: 0 > > > 8 headers, 0 lines > Message is NOTIFY > hm3*CLI> > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Martin Pycko > > Sent: Saturday, August 30, 2003 12:30 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Packet8 DTA310 > > > > Post the sip debug .. maybe someone will help you. > > > > Martin > > > > On Sat, 30 Aug 2003, Andrew Joakimsen wrote: > > > > > Has anyone been successful in using the DTA310 as provided by > Packet8 to > > > work with asterisk? I have gotten it to register with Asterisk but > > > whenever I try to dial a call all I get is silence, when I dial an > > > invalid extension I get a fast busy signal. When looking at the SIP > > > debug it seems that it is transmitting XML. > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vmail.cgi forward problems
I am testing out vmail.cgi I can listen to my messages, but I can't forward them to another user. I get the following error message: Software error: Invalid new mailbox That doesn't tell me much, so I'm hoping that somebody will be able to help me out. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unified Messaging Support ?
Do the ports have to be real (like XP100) or can you use the ztdummy (the dummy zapatel driver) for the ports required for hylafax? Lee Goodman - Original Message - From: "Michael Bielicki" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 4:18 AM Subject: Re: [Asterisk-Users] Unified Messaging Support ? > On Monday 01 September 2003 4:28 am, [EMAIL PROTECTED] wrote: > > On Tue, 1 Sep 2003, Tarun Banka wrote: > > > One quick question. Does anyone has experience implementing > > > unified messaging (UM) using Asterisk. Does Asterisk has support > > > for UM ? > > > > Not really. Half of it works. If you mean by UM (Voicemail integrated to > > your email box, and then send an SMS/Page etc to you) then its all doable. > > > > The problem happens when you want to bring fax into the picture. That > > cannot be handled currently, as in, * can't take a fax, and route it to > > your email as a PDF. > > > > But, you could use a third-party fax thingamajig and I'm sure connect it > > to * for a good UM solution. > Just pass it to hylafax and you fly, but it requires some planning cause you > will need ddouble the amount of ports plus the fax devices for hylafax > > > > -- > > wasim > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
This happens only on relaod. You can disable reload routine in chan_h323.c ... Martin On 1 Sep 2003, Michael wrote: > I'm running the CVS from last week and from day one (over 4 months now) > I've had this problem where asterisk core dumps when using chan_h323. > > It appears to be a problem with pwlib and the console, but I'm not sure > how to read the below output from gdb. I can start Asterisk just fine > and chan_h323 works great when sending and receiving calls. I only have > this core dump problem when sending a reload to Asterisk via the CLI or > "asterisk -rx "reload"". > > Environment paths: > LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib > PWLIBDIR=/usr/src/pwlib > OPENH323DIR=/usr/src/openh323 > > Core dump info: > (gdb) bt > #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, > PIntArray const&, PTimeInterval const&) () >from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 > #1 0x48315a2a in PSocket::Select(PSocket::SelectList&, > PSocket::SelectList&, PSocket::SelectList&, PTimeInterval const&) () > from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 > #2 0x483151a7 in PSocket::Select(PSocket::SelectList&, PTimeInterval > const&) () >from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 > #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper&, > H323RasPDU&, H323TransportAddress const&) > () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 > #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress > const&) () >from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 > #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress > const&) () >from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 > #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const&, > H323Transport*) () >from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 > #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, > gatekeeper=0x41efe680 "65.39.220.195", > secret=0x41efe700 "") at ast_h323.cpp:949 > #8 0x41eeed81 in reload () at chan_h323.c:1595 > #9 0x08055362 in ast_module_reload () at loader.c:159 > #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at > cli.c:105 > #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 > #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 > #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 > > Thanks, > > Michael > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message-waiting-indicator thru ZAP interfaces- how to?
On Tue, 2 Sep 2003, John Todd wrote: > >I'm trying to make the MWI indicators on my client's Vodavi Starplus > >DHS phones work. The actual signalling - in-band DTMF from the ZAP > >interfaces directly to the PBX system - works fine. I can manually > >tell asterisk to send "#9610" as DTMF and voila, the MWI on > >extension 10 lights (or goes out.) The question is, how is this > >integrated with voicemail, i.e. so that the MWI turns on and off > >appropriately, when new messages arrive and after a user has > >listened to their messages? > > > >I've checked the last two months of mailing-list messages but found > >no mention of this situation. Any tips or pointers to online docs > >would be appreciated. > > > >Thanks, > > > >Sam > > > >P.S. Thanks to Jsmith for the fast, simple answer to my last > >question re: version number in CVS not updating. > > I'm afraid that the answer to this, without programming some stuff > inside of Asterisk, is uuugly. I suspect it will involve using > perl or shell scripts to actually peek inside of the > /var/spool/asterisk/vm directories and check things manually, out of > a cron job or out of the "h" context with a System call. Then, a > call would be created by the script (see sample.call) - just > thinking about this method gives me the willies. > > The clean way would be to put a tiny call into the voicemail app (or > would it be in app.c?) that triggers an outbound call with the > appropriate parameters (see sample.call) I was working with someone over the weekend that is working on something like this, and it might even be the same type of system because the DTMF trigger looks similar. The easiest way to do this is with an external daemon that connects to the manager interface. You just need to watch for MessageWaiting events and when you see a change of state trigger an Originate action to dial out and enter the required DTMF. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
On Tue, 2003-09-02 at 10:00, [EMAIL PROTECTED] wrote: > php is not just a web scripting language anymore. it has been used in > other ways for quite a while now. it works nicely from the command line, > can be used with ncurses and with gtk. there are several well-known > respectable large projects out there built upon php. i usually find that > php's biggest critics are those who know the least about the language. > however that holds true with pretty much any technology. linux suffers > from the same type of critics. Just to point out, I am a php developer. I actually am employed to create and maintain a large webapp in php. I like the fact that I can take my php or perl scripts and not have to change much to them to work in the other language. Well if they are simple enough. There is enough well known documented problems with php. Just saying that because it is used in large projects doesn't change whether it is suited to the task. There are enough people on this planet, that statistically you will find enough people who refuse to admit the are using a square peg for the round hole. They eventually find a big enough hammer to make it "work". Once you have learned php, you only need learn a few more characters and you are ready to do simple perl programming. Then with time you are able to do very sophisticated things in perl. Hopefully it helps your php, otherwise it will annoy you when you go back and find the limitations of the php language. Eventually php will chase perl enough to get where perl is now. It will be a while as perl 6 still isn't due out for quite a while. > > On Sun, 2003-08-31 at 16:07, Josh Edwards wrote: > >> Are there any examples for ther psql or agi scriptscan I use php > >> with > >> agi > > > > Why do people try to shoe horn the wrong tools into this arena? You are > > better off using perl than php. Yes you can use php, but it is not meant > > to be used in a non web based applications. PHP is optimized for quick > > short run applications. Use perl, ruby, python, shell, or any other > > language that is intended to run systems and long run applications. > > > > Next point, there are already agi examples that have been included in > > the asterisk distribution. I have submitted psql extension logic here > > before. There is even documentation for it in the source code itself. > > Examples abound if you just look a little. They won't be exactly what > > you are asking for, but they will point you to how to do it yourself. > > > >> >From: Steven Critchfield > >> >Reply-To: [EMAIL PROTECTED] > >> >To: [EMAIL PROTECTED] > >> >Subject: Re: [Asterisk-Users] Newbie IVR question > >> >Date: 31 Aug 2003 15:54:53 -0500 > >> > > >> >On Sun, 2003-08-31 at 15:39, Josh Edwards wrote: > >> > > let me first say this is an amazing product. > >> > > > >> > > > >> > > ok here is my question > >> > > > >> > > what I want to do is be able to have people call me and answar > >> > > questions. The answars to there questions would need to be stored > >> in a > >> > > mysql database. > >> > > > >> > > so > >> > > > >> > > call comes in > >> > > > >> > > asterisks plays question > >> > > > >> > > asterisk waits for answar > >> > > > >> > > caller presses > >> > > 3 > >> > > > >> > > into a sql table goes (callerid,questionnum,3) > >> > > > >> > > on to next question. > >> > > > >> > > > >> > > Can this be done in asterisks, if so, can anyone point me in the > >> > > correct direction? howto's code examples etc > >> > > >> > > >> >If you are stuck on mysql, then you will need to do this in agi. If > >> you > >> >could do it in postgres, then there is the PSQL commands. > >> >-- > >> >Steven Critchfield > >> > > >> >___ > >> >Asterisk-Users mailing list > >> >[EMAIL PROTECTED] > >> >http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> __ > >> Get MSN 8and enjoy automatic e-mail virus protection. > >> ___ Asterisk-Users mailing > >> list [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting to an Ericsson AXT121 with a DigiumWildcat E100 card
Your configs look ok. All you need is BNC to RJ45 converter (I think the standard is G.703) regards Martin On Tue, 2 Sep 2003, Langley, Sean wrote: > Dear Telcotype Braniacs, > > I have tried doing a google search to find out what this switch looks like, what the > physical interface is, but havn't been successful. I am quite new to the ISDN world > so I'm not sure what to expect when I see this switch. Does anyone have any > experience connecting to this switch? The following is basically all I have been > given for info: > > 1) Ericsson, AXT121, interface ETCST (ROF 137 2764/2 rev R3A), Euro-ISDN. Connected > to FMV test network. > 2) Subscriber normally synchronised (sync) from network. > 3) 2,048 Mbit, 32TS-PCM, CCS in TS16. > 4) HDB3, 75 ohm coax. > 5) CRC check enabled. > > Like I said, I am new to the ISDN world, my digium card has an RJ45 connection type, > what do I need to physically connect to the above mentioned switch? > > I presume my zaptel configuration will be: > span=1,1,0,ccs,hdb3,crc4 > bchan=1-15 > dchan=16 > bchan=17-31 > loadzone=uk (not sure whether this is correct for sweden?) > defaultzone=uk > > and for zapata: > switchtype=euroisdn > signalling=pri_net > context=inc-e1 (or default...) > channel=1-15,17-31 > > Regards, > > Sean Langley, P.Eng > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RedHat Distribution
At 12:29 PM 9/2/2003 +0100, you wrote: I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend. Redhat 9 works perfectly. Install with the kernel sources and devel libraries, and the developers software, i.e. gcc, and upgrade to most recent rpms before making asterisk. --Ernest
Re: [Asterisk-Users] Low bit rate codec (speex)
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT) Brian West <[EMAIL PROTECTED]> wrote: > I opened a request on bugs.digium.com for this feature. The 6k and 8k > codecs are very impressive also. > > bkw > Where can I see the status of this request? []'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting to an Ericsson AXT121 with a Digium Wildcat E100 card
Dear Telcotype Braniacs, I have tried doing a google search to find out what this switch looks like, what the physical interface is, but havn't been successful. I am quite new to the ISDN world so I'm not sure what to expect when I see this switch. Does anyone have any experience connecting to this switch? The following is basically all I have been given for info: 1) Ericsson, AXT121, interface ETCST (ROF 137 2764/2 rev R3A), Euro-ISDN. Connected to FMV test network. 2) Subscriber normally synchronised (sync) from network. 3) 2,048 Mbit, 32TS-PCM, CCS in TS16. 4) HDB3, 75 ohm coax. 5) CRC check enabled. Like I said, I am new to the ISDN world, my digium card has an RJ45 connection type, what do I need to physically connect to the above mentioned switch? I presume my zaptel configuration will be: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk (not sure whether this is correct for sweden?) defaultzone=uk and for zapata: switchtype=euroisdn signalling=pri_net context=inc-e1 (or default...) channel=1-15,17-31 Regards, Sean Langley, P.Eng ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low bit rate codec (speex)
I opened a request on bugs.digium.com for this feature. The 6k and 8k codecs are very impressive also. bkw On Tue, 2 Sep 2003, Eduardo Goncalves wrote: > Hello, > > I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps. > With asterisk, what's the bit rate used by speex? Is it possible to have > asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to > implement? > > thanks in advance > Eduardo > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
php is not just a web scripting language anymore. it has been used in other ways for quite a while now. it works nicely from the command line, can be used with ncurses and with gtk. there are several well-known respectable large projects out there built upon php. i usually find that php's biggest critics are those who know the least about the language. however that holds true with pretty much any technology. linux suffers from the same type of critics. > On Sun, 2003-08-31 at 16:07, Josh Edwards wrote: >> Are there any examples for ther psql or agi scriptscan I use php >> with >> agi > > Why do people try to shoe horn the wrong tools into this arena? You are > better off using perl than php. Yes you can use php, but it is not meant > to be used in a non web based applications. PHP is optimized for quick > short run applications. Use perl, ruby, python, shell, or any other > language that is intended to run systems and long run applications. > > Next point, there are already agi examples that have been included in > the asterisk distribution. I have submitted psql extension logic here > before. There is even documentation for it in the source code itself. > Examples abound if you just look a little. They won't be exactly what > you are asking for, but they will point you to how to do it yourself. > >> >From: Steven Critchfield >> >Reply-To: [EMAIL PROTECTED] >> >To: [EMAIL PROTECTED] >> >Subject: Re: [Asterisk-Users] Newbie IVR question >> >Date: 31 Aug 2003 15:54:53 -0500 >> > >> >On Sun, 2003-08-31 at 15:39, Josh Edwards wrote: >> > > let me first say this is an amazing product. >> > > >> > > >> > > ok here is my question >> > > >> > > what I want to do is be able to have people call me and answar >> > > questions. The answars to there questions would need to be stored >> in a >> > > mysql database. >> > > >> > > so >> > > >> > > call comes in >> > > >> > > asterisks plays question >> > > >> > > asterisk waits for answar >> > > >> > > caller presses >> > > 3 >> > > >> > > into a sql table goes (callerid,questionnum,3) >> > > >> > > on to next question. >> > > >> > > >> > > Can this be done in asterisks, if so, can anyone point me in the >> > > correct direction? howto's code examples etc >> > >> > >> >If you are stuck on mysql, then you will need to do this in agi. If >> you >> >could do it in postgres, then there is the PSQL commands. >> >-- >> >Steven Critchfield >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> __ >> Get MSN 8and enjoy automatic e-mail virus protection. >> ___ Asterisk-Users mailing >> list [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installation Problem
On Tue, 2003-09-02 at 02:27, John Todd wrote: > Phil - >Here are my "generic" notes and reminders for Asterisk on Debian. > These may be hacks; your mileage may vary. > > debian asterisk install notes: > - in asterisk/Makefile: added "-I/usr/local/ssl/include" to CFLAGS line > - in asterisk/res/Makefile: added "-L/usr/local/ssl/lib" to CRYPTO_LIBS line > - in zaptel/Makefile: commented out KFLAGS+=-DCONFIG_ZAPATA_PPP line > - installed libnewt-dev > - installed newt-tcl (?needed) > - installed "apt-get source openssl" > - installed "apt-get install openssl" There are dev packages for openssl so the the source is not necessary. The dev packages put the headers in the right place and therefore the /usr/local/ changes aren't necessary. libnewt-dev is important so you can compile zttool and astman. newt-tcl shouldn't be necessary, but may be a debian dependency thing. The PPP line is only needed if either your kernel doesn't have PPP support, or if you don't plan on using the PPP options on the T1/E1 interfaces. Hope that clears up any problems. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn4linux
I noticed in the documentation that it is possible to use isdn4linux compatible hardware with *. I have a Dynalink 6692 PCI card which is ISDN4Linux compatible. How do I use it within * ? I would be grateful for any help. Thanks Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax with hylafax (changed subject)
But, you could use a third-party fax thingamajig and I'm sure connect it to * for a good UM solution. Just pass it to hylafax and you fly, but it requires some planning cause you will need ddouble the amount of ports plus the fax devices for hylafax Interesting - please, do you have time to elaborate a bit more on that functionality? What kind of ports needs to be doubled? Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] RedHat Distribution
Redhat 8 and 9 have both worked fine for me.. I have an install guide at http://members.lycos.co.uk/wipe_out/asterisk If you are intersted.. Later- Original Message -From: "Francisco Mesquita" <[EMAIL PROTECTED]>Date: Tue, 2 Sep 2003 12:29:23 +0100To: <[EMAIL PROTECTED]>Subject: [Asterisk-Users] RedHat Distribution Hi, I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend. Best regards, Francisco Mesquita -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Low bit rate codec (speex)
Hello, I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps. With asterisk, what's the bit rate used by speex? Is it possible to have asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to implement? thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RedHat Distribution
Hi, I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend. Best regards, Francisco Mesquita
Re: [Asterisk-Users] H.323 Support
chan_h323 is built into asterisk. Check the /usr/src/asterisk/channels/h323 directory for more info. - Original Message - From: "Phillip Britt" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 1:12 PM Subject: [Asterisk-Users] H.323 Support Hi, I am currently using Asterisk and want to add H.323 support for talking to our gateway routers, which use gnkgk Is the package "Asterisk-oh323" the right thing to use, or are there better ways of achieving h.323 support in Asterisk. Thanks, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 Support
Hi, I am currently using Asterisk and want to add H.323 support for talking to our gateway routers, which use gnkgk Is the package "Asterisk-oh323" the right thing to use, or are there better ways of achieving h.323 support in Asterisk. Thanks, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
I tried specifying rxgain & txgain, copied the format some some message on asterisk-users Result was asterisk bombed out & didn't even load due to not being able to understand the config file .. what's the exact syntax that works?? cheers Dave - Original Message - From: "Fredrik Hedberg" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 9:21 AM Subject: Re: [Asterisk-Users] SIP and ECHO > What have you specified as rx and txgain? > > Simon McAuliffe wrote: > > >I've been having the same problem too, except for me it only happens > >occasionnally. > > > >I'm not 100% sure of this, but it seems that for very local calls (eg across > >the city) I never get echo. For calls that go longer distance (say 500km or > >more), or through some closer call centres, I'm getting the echo. I don't > >get the echo on an analogue POTS connection to the same places (it is > >clearly only happening on our asterisk system). > > > >This might indicate some link between echo cancellation and delayed audio, > >but if so, its sensitive to very small delays. > > > >The echo can only be heard at our end, there is no trace of it at the other > >end. > > > >I'm using ATAs doing SIP to Asterisk and through a PRI connection to a > >Telco. Echo cancellation is turned on and showing as activated on the Zap > >channels. Echo cancellation is also enabled on the ATAs. > > > >- Original Message - > >From: "Brian J. Schrock" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Friday, August 29, 2003 3:16 AM > >Subject: [Asterisk-Users] SIP and ECHO > > > > > > > > > >>Hello, > >> > >>I have read the information on echo and SIP in the FAQ and I have > >>scoured the mailing list for possible solutions, but as yet I have not > >>been able to get rid of this echo. > >> > >>I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > >>into an asterisk server. If I call between the Sip Phone > >>(Budgettone-100) and the 4 FXS ports everything sounds great. If I call > >>out to the PSTN through the FXO cards I get horrible echo, I have even > >>been able when talking loud enough to get a horrible feedback loop > >>going. I have tried 4 different echo cancellers in the Makefile for the > >>Zap drivers and nonoe of them changed the situation. > >> > >>I have echocancel = (Any where from 1 - 256, I have tried alot of > >>different values), and I have echocanelwhenbridged = yes.I only hear the > >>echo start when the call gets bridged onto the outgoing PSTN lines. > >> > >>Is there anything I can do? > >> > >>Brian J. Schrock > >> > >>___ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > Fredrik Hedberg > > Telavox ABDirect: +46 46 6220013 > Lilla torg 1Phone: +46 46 622 > S-211 34 MalmoMobile: +46 70 3323033 > SwedenWeb: www.telavox.se > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
On 01 Sep 2003 23:23:53 -0500 Steven Critchfield <[EMAIL PROTECTED]> wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* On Mon, 2003-09-01 at 22:23, Dave Packham wrote: http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * And the benefit of using a commercial software that costs money is ? I just love the fact that they claim stunning sound quality when all the variables are outside it's control. Software doesn't make you sound card better. The codecs are standard and therefore not going to be improved by this software. Not to mention it only runs on windows, and it's minimum requirements is higher than that needed for an asterisk system. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users C'mon Steven, give the poor Nero folk a break. I'm sure they have marketing drones that they have to keep busy coming up with this 'stunning sound quality' crap. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gnuGK + h323 Caller ID
For some reason, chan_h323 ignores the callerid and puts your IP address in instead. I've modded my chan_h323 to use the caller's id instead. Trival change but I'm guessing there's a reason why it isn't so in the first place. Anyone know why? Adam Hart - Original Message - From: Rattana BIV To: [EMAIL PROTECTED] Sent: Monday, September 01, 2003 7:19 PM Subject: [Asterisk-Users] gnuGK + h323 Caller ID Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. - Original Message - From: "Brian J. Schrock" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO > Hello, > > I have read the information on echo and SIP in the FAQ and I have > scoured the mailing list for possible solutions, but as yet I have not > been able to get rid of this echo. > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > into an asterisk server. If I call between the Sip Phone > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > out to the PSTN through the FXO cards I get horrible echo, I have even > been able when talking loud enough to get a horrible feedback loop > going. I have tried 4 different echo cancellers in the Makefile for the > Zap drivers and nonoe of them changed the situation. > > I have echocancel = (Any where from 1 - 256, I have tried alot of > different values), and I have echocanelwhenbridged = yes.I only hear the > echo start when the call gets bridged onto the outgoing PSTN lines. > > Is there anything I can do? > > Brian J. Schrock > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. - Original Message - From: "Brian J. Schrock" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO > Hello, > > I have read the information on echo and SIP in the FAQ and I have > scoured the mailing list for possible solutions, but as yet I have not > been able to get rid of this echo. > > I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed > into an asterisk server. If I call between the Sip Phone > (Budgettone-100) and the 4 FXS ports everything sounds great. If I call > out to the PSTN through the FXO cards I get horrible echo, I have even > been able when talking loud enough to get a horrible feedback loop > going. I have tried 4 different echo cancellers in the Makefile for the > Zap drivers and nonoe of them changed the situation. > > I have echocancel = (Any where from 1 - 256, I have tried alot of > different values), and I have echocanelwhenbridged = yes.I only hear the > echo start when the call gets bridged onto the outgoing PSTN lines. > > Is there anything I can do? > > Brian J. Schrock > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample configs
Hi, Ok, the phones are working and seem to be loading the correct info from the tftp server. However, I am unable to make them perform any functions (calling another extension, going to voicemail, etc.). I do not have any telephony interface installed yet, only a single ethernet card. Do I need to install the ztdummy driver to make any of this work? And if so, how do I do that? Thanks, Travis Microserv At 07:52 PM 8/29/2003 -0700, you wrote: Travis Johnson wrote: Hi, We are just getting started setting up an Asterisk VoIP server. We are very experienced with Linux, networking, tcp/ip, etc. However, some existing sample config files for using Cisco VoIP phones with this server would be VERY helpful. Thanks, Travis Microserv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This is a bare minimum assuming you start with a phone will all default config settings. /etc/asterisk/sip.conf == [1234] callerid="Your Name" <1234> context=yourcontext type=friend secret=yourpassword host=dynamic defaultip=youraddress mailbox=1234 /tftpboot/ (presumably, where your phone config files are) == SIPDefault.cnf == proxy1_address: "asterisk.ip.address" proxy_register: 1 SIPyour:mac:address.cnf === line1_name: 1234 line1_authname: "1234" line1_password: "yourpassword" line1_displayname: "Your Name" ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
I downlaoded it and tried it, SIPPS. Nice featureful sip client, however, I haven't been able to get it to pass dtmf to *. I don't know if this is a software restriction or not, but I have emailed nero asking them for their opinion of this, as it is, in my case, a LARGE restriction when trying to deal with IVR's, and esp. * voicemail. - Original Message - From: "Dave Packham" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]> Sent: Monday, September 01, 2003 10:23 PM Subject: [Asterisk-Users] Sip Software from Nero Folk? > http://www.nero.com/us/631911127302064.html > > > Have you all seen this? > > Its a SIP softphone put out by the people that do the CD burning software Nero... > > Check it out it works with * > > Dave > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
On Mon, 2003-09-01 at 22:23, Dave Packham wrote: > http://www.nero.com/us/631911127302064.html > > > Have you all seen this? > > Its a SIP softphone put out by the people that do the CD burning software Nero... > > Check it out it works with * And the benefit of using a commercial software that costs money is ? I just love the fact that they claim stunning sound quality when all the variables are outside it's control. Software doesn't make you sound card better. The codecs are standard and therefore not going to be improved by this software. Not to mention it only runs on windows, and it's minimum requirements is higher than that needed for an asterisk system. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gnuGK + h323 Caller ID
For some reason, chan_h323 ignores the callerid and puts your IP address in instead. I've modded my chan_h323 to use the caller's id instead. Trival change but I'm guessing there's a reason why it isn't so in the first place. Anyone know why? Adam Hart - Original Message - From: Rattana BIV To: [EMAIL PROTECTED] Sent: Monday, September 01, 2003 7:19 PM Subject: [Asterisk-Users] gnuGK + h323 Caller ID Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unified Messaging Support ?
On Tue, 1 Sep 2003, Tarun Banka wrote: > One quick question. Does anyone has experience implementing > unified messaging (UM) using Asterisk. Does Asterisk has support > for UM ? Not really. Half of it works. If you mean by UM (Voicemail integrated to your email box, and then send an SMS/Page etc to you) then its all doable. The problem happens when you want to bring fax into the picture. That cannot be handled currently, as in, * can't take a fax, and route it to your email as a PDF. But, you could use a third-party fax thingamajig and I'm sure connect it to * for a good UM solution. -- wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Software from Nero Folk?
http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change include contexts runtime
On Monday 01 September 2003 03:51, Mickey Binder wrote:
> How do I change the dialplan runtime, if I for example wants all
> calls on the main number to be answered by a voicemail (when it is
> out-of-office hours).
> I want to be able to change the configuration by pressing a DTMF
> combination e.g. *82. Can't figure out whether it is necessary to
> change contexts or how to do it.
>
> I have read a lot of examples and config documentation, but I can't
> figure out how to do it.
>
> I know there are commands from the CLI to include and not include
> contexts but I can't get them to work.
> If i write 'include context in default' I can see by 'show dialplan'
> that 'context' is included in default. But if I want to include a
> context named office by typing 'include office in default' I just get
> 'No such command 'include office' (type 'help for help)
Use the DB routines and GotoIf. Example:
exten => 999,1,DBPut(mystore/isopen=1)
exten => *82,1,DBPut(mystore/isopen=0)
exten => s,1,DBGet(amiopen=mystore/isopen)
exten => s,2,GotoIf($[${amiopen} = 0]?closed|s|1)
Obviously, you'll want to put the extensions that turn the system on and
off in a context which is not referenced by incoming calls.
-Tilghman
___
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Re: [Asterisk-Users] Filling PHP Variable from EXTENSION in AGI
Brancaleoni Matteo Thank you very much for your pointers. I wrote a little PHP function which read an input from http://stdin I can extract it and choose a needed value. Now a variable of PHP-based-AGI script contents a dialed extension :) Romsun Pramudito Brancaleoni Matteo <[EMAIL PROTECTED]> wrote: > Il sab, 2003-08-30 alle 22:40, romsun p ha scritto: > > Hellooo... > > hi > > > > Is it possible to fill a variable of PHP-based-AGI-script > > from dialed extension ? > yes > > > > This is what I need to achieve: > > If someone dial an extension, say 777, > > I want the dialed extension (777) be filled into > > PHP variable. I need the dialed extension become > > a condition of PHP script. > > as soon as you start the script with AGI(scriptname), > asterisk sends out some vars via std input to the script. > just read that output from the script, parse it and you'll > have your dialled extension, along with other vars. > See app_agi.c to get what vars are sent by asterisk to the agi > script. > agi_extension is what you need (contains the dialled number) > > I wrote a little function in php that reads this output, parse > it and put it into an assoc array, so I can have all > the vars in a single & easy array. > > > > > > Help please... > > Thanks > > > > romsun > > _ > > This mail sent using V-webmail - http://www.v-webmail.org > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users _ This mail sent using V-webmail - http://www.v-webmail.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming phone dialing / IXJ
Hi all, Does anyone have a working IXJ / Dial in config they'd lke to share with me? Thanks, Andrew Griffiths ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Actually, I do have that. i've tried inband, as well as rfc2883. Neither work. I'm going back and forth with ahead software on the issue, and they're doing a little bit of looking into it. Doesn't even work when clicking on the numbers, as required by the software, as someone else pointed out, that was an obvious "feature" i noticed right off the bat. -Josh - Original Message - From: "Gavin Hollinger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 1:58 AM Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > haven't been able to get it to pass dtmf to *. I don't know if this > > Do you have > dtmfmode=inband > in sip.conf? > > http://www.sippstar.com/en/631927444894185.html > > Q.: DTMF generated by SIPPS is not recognized by other > applications. > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > application awaiting an event instead of a tone will not be able to work > with SIPPS > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Change include contexts runtime
-Original Message- From: Tomas Prybil [mailto:[EMAIL PROTECTED] Sent: 2. september 2003 10:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Change include contexts runtime Mickey Binder wrote: >>It looks like it. With DBput and DBget im able to change the variable values >>and then branch to different contexts with GotoIf. Now I just need to >>implement the right logic for the different situations. > >> >> >And maybe be able to get some sort of feedback to the users. >Change of dialtone or visual indication? > > >/t > Yeah...I thought of making a voice response telling the user whether he turned "out-of-office" voicemail on or off, and then hangup afterwards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change include contexts runtime
Mickey Binder wrote: It looks like it. With DBput and DBget im able to change the variable values and then branch to different contexts with GotoIf. Now I just need to implement the right logic for the different situations. And maybe be able to get some sort of feedback to the users. Change of dialtone or visual indication? /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
