Hi!
> I've suffering cut offs after 6 or 7 seconds a call is answered,
> incoming calls are working fine, but outgoing ones show the gollowing
> messages when are being dropped
> [...]
> It seems the SIP ACK is not being received properly.
I can confirm this issue: In my case it happens with cal
Hi!
> > Side note: Stay away from solutions that use mISDN, instead go with
> > Zaptel (DAHDI), Woomera or CAPI.
>
> Interesting.
>
> I've been usng mISDN for some years now without issues. Why should I
> migrate to DAHDI?
None - if you are happy then don't touch it. :-) Otherwise search this
Hi!
> > 2. Add BRI card(s) to the computer to run Asterisk and somehow hook
> > up the Samsung.
>
> Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the
> asterisk box. But then, might as well dump the Samsung and just put
> VoIP phones on everyones desks.
If you decide to go
Hi!
> It is causing an issue for me. One SIP UA works fine - ring, forward, etc.
> While the other does not.
Make the UAs listen on different ports (for example 5060 and 5062) and
see if that solves your problem - if you can't make them have different
IPs, that is.
Also be sure to fully under
> exten => s,n,Set(vgLabel=vg(${number}+1))
> exten => s,n,GoTo(${vgLabel})
>
> But in stead of vgLabel becoming the SUM of 2 numbers, it is just a
> string :
Use the MATH function.
Philipp
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Hi!
> [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
> <6839>, digest has <3169>
You most likely have two SIP UAs that use the same IP, of which the 6839
account is listed last in sip.conf while 3169 is trying to auth
(unsuccessfully).
Philipp
--
___
Hi!
> I have asterisk 1.4
> I want to make a MGCP trunk as a client to connect to a provider who is
> using MGCP protocol, he provided me with user & password,
You will most probably need 1.8 for this, with 1.4 you will certainly not
be able to succed. Read more:
http://www.voip-info.org/wiki/v
Hi!
> I've turned off t.38 and all of the codecs except ulaw; I still have the
> same problems. SOMETIMES it works. Other times, the sniffer clearly
> shows that the media simply isn't being sent. NOTHING is being sent.
>
> Anything else I should check?
Look at the firewalls and possible SIP
Hi!
> In this case I will want to use Snom phones. TFTP is available, but no FTP
> (with indeed then a username and password). FTP would be great...
You could also consider to use the "SNOM Redirection Service" for
provisioning:
http://wiki.snom.com/PROVISIONING
Remark: TR-69 provisioning do
Hi!
> but all of a sudden we have all calls origination from one sip
> extension opening channels which have the name of another sip extension
> in the channel name.
Do the devices of this extension happen to have the same IP address?
Philipp
--
__
Hi!
> To complicate it a bit, is there a way to extract the rejection reason (in
> this case 486) without using the DIALSTATUS and to use it? It can be
> useful for rejections that are not handled by DIALSTATUS.
If the other side is also an Asterisk system: When call is hang up that
involves a S
Hi!
> Is there a way/software which can act as a middle man between asterisk
> and the ethernet ports, and by default sends registrations to asterisk
> only from eth0, and if this port fails, sends registration coming in
> from eth1?
Spanning Tree (STP, RSTP, MSTP)
--
Hi!
> Do I need a low REGISTER timeout value for this to work ? Something
> like 60seconds, so the IP-phones register every 60 seconds...
This can also be done without SRV records if you have phones that provide
a fallback SIP registrar entry - snom for example does, and many other
vendors do
Hi!
> Trunking only reduces overhead after 4+ calls, so that shouldn't help
> either. (Since this occurs at 2 calls)
Trunking requires a timing source, and you might have trouble with your
timing, that is why I suggested this (and because you did not tell us
wether you have trunking enabled or
Hi!
> Can someone suggest where to look? Could this be the ITSP?
- turn off IAX trunking mode
- test with SIP to find if it IAX causing the trouble
- capture the RTP traffice on the other side and let wireshark have a
look at that stats (loss, jitter)
Philipp
--
__
Hi!
> I'm having difficulty with registering a SIP account in a Snom 320 IP-
> phone.
Do a SIP trace on your SNOM phone, and you will most probably see that
the 401 reply of Asterisk does not arrive on the phone. Then check your
STUN/ICE settings on the phone in combination with the nat= settin
Hi!
> Can you tell me how I can get my Snom 320 auto-answer the call when I
> use the Page()-command ?
Configure a special identity on the SNOM that is set to auto-answer in
the phone's configuration. Or consider to use multicast instead of Page()
if your network topology doesn't stand in the
Hi!
> Just out of interest, have you ever got this working?
Yes, sure.
> Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono
> 16 bit wav files is a bit dodgy
Very well possible. Also look at the individual "identity x"
configuration and consider to select "Custom ringtone",
Hi!
> I've also been lead to believe that I can set a URL for a custom
> ringtone in the Alert-Info header but can't find the exact syntax
> anywhere.
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+snom
http://asterisk.snom.com/index.php/Asterisk_1.4/Ringtone_Configuration
Philipp
Hi!
> Looks like I still don't understand how SHARED works :-(
>
> exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Please check if in Asterisk 1.6 the Syntax for passing arguments to the M
option of Dial() has changed.
> [macro-test]
> exten => s,n,Set(SHA
Hi!
> Well, only problem I see, is to how pass channel name from macro to h
> extension...
SHARED() or CDR(userfield)
Philipp
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Hi!
>> Why is it a problem? It sounds like Asterisk does silence suppression.
>
> 1) With no rtp traffic, the nat device will drop the connection in it's
> nat table and thus disconnecting the softphone from Asterisk. (after
> the router's timeout period of course)
>
> 2) The other issue is you
Hi!
> traffic to an IP address - then, rather than me manually analysing with
> wireshark, will analyze the cap file and produce stats on jitter, lag,
> delta etc.
This is what RTCP was made for.
Philipp
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Hi!
> There are 2 things I can't understand
> - 1. how can I know channel name?
${CHANNEL}
> 2. where should I call this SHARED function? before Dial, after Dial?
Either In the macro that you specify using the M option of Dial() or in
the h extension. You will, however, have trouble treating t
Hi!
> > And the third hit in my google result is this:
> >
> > http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
> >
> > Since I mentioned in my previous message that you will find the answer
> > in the archive of this list you could have found that even without
> > google. gm
Hi!
> I need the system to be resilient to any network partition, so that
> anyone can send announces from any mic to all the reachable clients.
> I'd need also to page a subset of all the speakers.
Most of the major phone vendors (that are employed by the users of this
list) have support for m
.slin is not .wav
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asterisk-users mail
Hi Dmitry!
> > Have you considered using Google (or your favourite search engine)?
>
> Shure, I searched and find nothing.
> > The search terms "C" will surely help you, and in
> > fact point you to the very archive of this mailing list.
Don't know where this quote comes from, but "C" is absolu
Hi!
> I see. I want to use SHARED function!
> Do you have example how to
> "to export them to the local call leg/channel "?
Have you considered using Google (or your favourite search engine)?
The search terms "asterisk function shared" will surely help you, and in
fact point you to the very arch
Hi!
> Could somebody tell me how to use SHARED function?
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared
> I want to get RTCP stats from SIP, but current channel is DAHDI.
> How can I get SIP channel?
If you have one DADHI and one SIP channel bridged together, then only for
Hi!
> notifyringing = no ; Control whether subscriptions already
> INUSE get sent RINGING when another call is sent (default: yes)
>
> Does this mean that when I mark this as "yes", a phone that already has
> taken a call will be send a second and third call ?!
No, not directly: This setting is
Hi!
> Does this shine new light to the problem ?!
No. Once more: Go and read doc/backtrace.txt.
And check if you have any meaningful information in /var/log/messages for
the timestamp when asterisk crashed.
Philipp
--
_
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Hi!
> DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
> work for me in asterisk 1.4.35. Any other method function to check the
> channel status
There is a backport available for 1.4:
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
I assume that with "does
Hi Jonas!
> It indicates to be a binary file, however I have not found instructions on
> dealing with this @ the link you gave me.
>
> Can you give me instruction on how to handle the core.pid file ?
Could I ask you again to make an effort to reduce your number of daily
postings to this list? I
Hi!
> I know I post a lot concerning this issue, but this is because this
> problem occurs on a production system and I feel very hot breathing down
> my neck.
Why not reduce the pressure and revert to 1.4.30 for the production
system until you have figued out the issue? That will give you more
Jonas,
everyone here supports you in your effort to get a good Asterisk
installation going, but could you ... maybe restrain yourself a little
bit and reduce the number of hasty postings you are sending to this
mailing list?
Thank you,
Philipp
--
Hi!
> I am running asterisk ver 1.2.4 and have faced this error:
Try a downgrade to Asterisk 0.7.1 ;->
Philipp
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Hi!
> is their a way to keep having a dialtone for the calling party when the
> macro is executed ?! Or not ?!
Consider using either FollowMe() or using the G option of Dial() instead.
Philipp
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Hi!
>> Yes, typically there is only one SMSC that can send you SMS on a fixed
>> line; look at its Caller ID to identify a SMS call.
>
> Even when the call is coming from a cellphone ?
A SMS is not really a call (at least not in the mobile world), and the
cellphone cannot directly send a SMS
Hi!
> 1. Do you have any experience with receiving incoming SMS on an analog or
> ISDN landline ? How can then you differentiate an SMS call from a voice
> call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems
> the way to tell an inbound call is an SMS one is to read the caller
Hi!
> We're using firmware 7.3.30 on an installation of Snom 300 phones.
> Should we stick with it, or do the newer firmwares have better support
> for Asterisk?
So what is it that you are missing that firmware 8 does offer? 7.3.30 is
rather stable and therefore a good choice.
Actually I woul
Hi!
> After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
> finishes. On the Asterisk console, I can see that the sound file is indeed
> playing, but we can't hear it. [...]
>
> I have tried so many things that I have lost count, and I humbly ask the
> collective intelligen
Hi!
>> My question is this. Is it possible to tell Asterisk to execute part
>> of a macro as a block without allowing any other commands to be
>> processed during that time?
>
> What would be a correct way to do this in 1.4.x?
*CLI> show application MacroExclusive
Philipp
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_
Hi!
> * Remove current STUN support from chan_sip.c. This change removes the
> current
> broken/useless STUN support from chan_sip.
> (Closes issue #17622. Reported by philipp2.
> Review: https://reviewboard.asterisk.org/r/855/)
What you do not see mentioned here, and that is a b
Hi!
> How can I remove the "Playing digits" from parkcall application?
In general you can address problems like this by creating your own set of
sounds files where the obstructing files are either simply missing or
replaced by silence. Use Set(LANGUAGE) right before the action (here:
parking t
Hi!
> > By a mixed environment I mean some Asterisk servers running on AMD and
> > some running on Intel
>
> If it was possible for that to matter, then the software would be very
> poorly written indeed. As another poster said, the only way that would
> have any effect is if you compiled binari
> read the value of var ${HANGUPCAUSE} next line to dial command.
So how can you be sure this has been populated by PRI and not by SIP?
This will not tell you which side hung up.
Philipp
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Hi!
> Is there ANYWAY to find out which party hang-up the call or if the call
> was cut-off due to other reasons?
The only way - apart from putting DAHDI or SIP into debug mode - I can
think of is to use the 'g' option of Dial(): If the remote side hangs up
then 'g' will come into effect; if t
Hi!
> Does anyone know if the behavior of 'r' has changed but was not
> documented? If yes, then how does one inject ringback audio before the
> call is answered on the called end?
Search this list for "progress" or "progressinband", and look at the voip-
info wiki.
http://www.voip-info.org/wik
> Only when I configure my Grandstream to use only G726 (I have 8
> choices), I see that the g726-codec is used.
> When I configure 7 x g726 and 1 x alaw, then again alaw is used !
>
> Is it normal that Asterisk has such a great preference for alaw ?! The
> moment the peer suggests codec alaw (ev
Hi!
> Although my previous posts in this forum have not received satisfying
> answers, here is another question from me.
You might want to consider to reqest a refund. ;->
> my question is can i use ChanIsAvail function to get the status of a user
> in the format SPI/user-id if i provide user i
> You might be able to record these snippets then pass them through the
> Vestec or Lumenvox Speech engine to get what you want.
Unfortunately that won't work because:
* the containing recordings/feeds can be quite long, can be
embedded/surrounded by silence, ringing tones, music or special ton
Ok, here's the challenge:
I would like to be able to find, match - and then react - upon prompts
that are presented by the outbound/remote side of a call. Think mobile
phone and "This user is temporarily unavailable".
Collecting a limited number of known prompt snippets should not be a
problem
> The mapping in Asterisk 1.4.24 is the problem: 402 "Payment Required"
> is mapped to 16 "Normal termination" instead of 21 "Call Rejected".
> Could you direct me to the relevant file of code where these mappings
> are done? Before reporting a bug I would like to confirm whether this
> issue has b
Hi!
>>> Let's say I call by SIP/trunk1/number and the proxy server is
>>> down, is there a way to getCHANUNAVAIL?
>> *CLI> core show application Dial
> Unfortunatelythe timeout parameter will not do the job for me. I need
> somethingequivalentto "qualify" to monitor the outboundproxy.
Why not
Hi!
> In the [general] section of sip.conf I have :
>
> disallow=all
> allow=g726
> allow=alaw
> allow=g729
> allow=gsm
So change the order there and see what happens.
> > * look at the variable SIP_CODEC for the inbound (first) call leg, and
> > in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OU
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
For quick testing to see if the codec works at all: Configure your phones
to do g726 only (so n
Hi!
> Question 1 :
> [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
> audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
> why is combined alaw|g726 and not g726|alaw (reverse) ??
Guess: Here the order presented has no meaning for the order of codec
Hi!
> Is there a way to change the mappings of disconnect reasons to certain
> SIP messages? E.G. I need to change the mapping for SIP 402 "Payment
> Required" from 16 (normal termination) like it is in 1.4.24 to 21
> (call rejected) as defined in RFC 3398.
* if you think the mapping is wrong, th
Hi there!
David has put up a patch to fix the STUN issues that has plagued Asterisk
1.6 ever since that feature was introduced. Now we need testers to verify
the patch, so please grab the patch (or check out the SVN branch) and add
your comments:
https://issues.asterisk.org/view.php?id=17622
Hi!
> i want to get channel-id before dialing so that i can dial using that
> channel id.
I am afraid that is not going to work. Maybe you should take a step back
and describe what it actually is that you are trying to accomplish.
Philipp
--
_
Hi!
> I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I
> change the fields name of database?
You will want the adaptive CDR backport to Asterisk 1.4:
https://issues.asterisk.org/view.php?id=1
http://svncommunity.digium.com/view/tilghman/branches/1.4/
Philipp
--
_
> We are running asteriskNow 1.4.18 and after a few days it becomes
> unresponsive and inbound INVITEs timeout.
Search this list for "DNS".
Philipp
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Hi!
> I'm working for Zoiper, you can contact us directly on supp...@zoiper.com
> Zoa
>
> > I will do a test call from a soft phone to my mobile. I can speak into
> > my headset and the audio is heard instantly. But if I speak into my
> > mobile there is a 1-2 second delay in the Audio. I am usin
Hi!
Three notes:
* as others have already mentioned: personally I would not Dial() from
within AGI using EXEC, but rather set extension and context and then let
the dialplan handle the Dial, and therefore complete that AGI before the
Dial; then possibly run another AGI after the call in the h
Hi!
- upgrade to a current 1.4 version, 1.4.17 is very old (you probably run
this because of the zaptel --> dahdi change, but still)
- do you have a SIP proxy or any "SIP-aware" hardware in your network
that might play tricks on you, e.g. a SIP ALG (application layer gateway)
on your Internet
Hi!
> 7) if john doe want to speak with caller assistant bridge the two
> lines using the transfer function of GXP2000 phone (REFER).
>
> After the transfer in the CDR i can't see the callerid of the caller,
> only data of the bridged call is reported.
>
> Any idea on what i can do to keep it ?
Hi!
> Great, but how exactly do i find that channel - that is my question -
> which command.
For the third time: Use the M option to Dial() and create a Macro. In
that macro use the SIPCHANINFO() or CHANNEL() function to get what you
want to get. No AGI (and AGI is a protocol while Ruby is a la
> The problem we are having with Asterisk is when we initiate a call via a
> Zap line and it goes out on a Sip line. When it goes out via Sip we hear
> no sound until the party we are calling answers the line.
Search for "progress" and/or "progressinband".
--
___
Hi!
> Depending on the version of Asterisk you are running you can call a macro
> or an agi as option to dial. These will be called when the line is
> answered and you can find the channel name of who answered.
Do as he says, look at the M option to Dial.
Philipp
--
__
Hi!
> I've worked with these before. They are designed to run a whole
> hospital shift, so there should be no worries regarding the battery.
Sounds good. The speaker phone quality is acceptable (the speaker is
quite small and points forward, not upwards in the direction of the ear),
or would y
Hi!
>> I've been asked to implement the following transfer workflow in an
>> asterisk system, and I'm not seeing an easy way to do the bolded steps
>> below (steps 4 and 5 for those with a text-only email client):
> You could create a dynamic meetme room for the 3 legs and drop out when
> done.
Hi!
> I´ve seen them at trade shows, I think I remember it being proprietary.
> What about using Dect handsets?
That Star Trek device has always interested me. Too bad they chose WiFi
over DECT, though.
Vocera badge:
* WLAN b/g
* Talktime 2-2.5 hours, standby 20-27 hours
* headset jack
* OLED d
Hi!
> I got some reports of (Debian Testing/Unstable) systems where the
> timerfd timing didn't work properly and the workaround was reverting to
> the pthreads one. I have not yet managed to reproduce them here.
>
> I wonder if this is the issue.
How about this:
http://svnview.digium.com/svn/as
Hi!
> > I'm experiencing a problem with my SIP channel's. When I have an
> > external connection for one of my SIP carrier's, I can listen to the
> > client and the client listens to me normally. The problem is when I will
> > transfer this connection, the call is mute for the extension I have
>
Hi!
> 2. my SOX (1.14.0) on CENTOS doesn't handle alaw files.
It surely does, only that you need to tell it explicitely to:
Use "-t ul" or "-t al" and you are fine.
Philipp
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Hi!
> client listens to me normally. The problem is when I will transfer this
> connection, the call is mute for the extension I have transfered. Only the
> client hears normally.
I *think* there is/was an entry in the bug tracker on this. You might
want to search https://issues.asterisk.org (al
Hi!
> Nobody uses chan_local
Absolutely nobody. Except you. ;->
Maybe this will help you: Search for "Asterisk timing", consider to not
run Asterisk in a virtual environment, and do not run X on the same box.
Makre sure to turn off silence suppression in your SIP client(s).
Search for "c
Hi!
> I am working on calling 2 registrations of same user on 2 different ip or
> ports. It works fine and both phones ring simultaneously. the problem is
> that there is one way audio, calling party can hear me but i can't hear
> calling party.
You need to make sure that these two phones use *di
Hi!
> 'Fax for Asterisk' is a commercial application sold by Digium. This is
> not their official support channel. Since you paid for the product, why
> not contact them directly about your problem?
Maybe because having to deal with Digium support is an ... uncomfortable
experience that I've ma
Hi!
> Is there a way to log the negotiated codec that was used for each call
> in CDR or in a separate log file?
Use CHANNEL(audionativeformat) - and do the same with the help of the M
option to Dial() for the remote call leg. Store that info in the CDR
"userfield", or create your own field if
> SEND >> 0.0.0.100:5060
?!
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asteris
Hi!
> >Provided the user doesn't have access to the firewall (eg. corporate or
> >hotel), and the firewall doesn't allow dynamic port opening through UPnP
> >or NAT-PMP...
>
> For those interested, I was tipped through private e-mail about using
> OpenVPN to open a steady tunnel between the clien
> When I dial a peer with no registration, SIP channel currently waits
> many seconds before returning ${DIALSTATUS} "CONGESTION" - how can I
> shorten this timeout?
Look at qualify=yes for that peer.
Use ChanIsAvail() before you dial.
Use SIPPEER(peername|status) to check registration status.
Use
Hi!
> Has anyone had experience installing it?
> yum install asterisk-chan_misdn
> I'ts the latest Trixbox Distro version and same issues exists if add in
> the Trixbox repo. FAILS as per below
Please search this list for recent messages on mISDN, or Google it.
You will find that mISDN v1 doe
Hi!
> The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
> protocol, and PtMP is an extension of its logic to make ISDN (BRI) phones
> behave somewhat like analog phones: allow you to connect several of them
> on the same line.
In other words:
While you *must* have exactly on
Hi!
> The network setup is :
> analogue+GXW / softphone --> Linksys WAG160N --> Asterisk server --> ITSP
> --> other networks
Do it step-by-step: Take the Asterisk server out of the equation, i.e.
call the destination directly with your softphone or the Grandstream ATA
and see if that removes t
Hi!
> Sometimes there is a long gap between Asterisk starting and devices
> being able to register.
First you should check your DNS setup - it has been discussed many a
times on this list.
Philipp
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Hi!
> For codecs use CHANNEL function, but you will only get CallLegA
> codecs. Without hacking Asterisk, you will not be able to get CallLegB
> codecs. Patch for Asterisk 1.4.33.1 attached to get such info.
Thank you! In the meanwhile I found that with the help of the M option to
Dial (macro c
Hi!
> Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
Most probably - who on this list would you like to test it for you? ;->
Philipp
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Hi!
> Because the codec is already chosen before the call is made, and you
> told it that g722 is permitted.
>
> There are all sorts of discussions in play about codec negotiation,
> but at this point in time, if you want different behaviour you'll need to
> go and code it yourself
Look at the l
Hi!
> Do you already have script to capture user's IP address? If not, check
> it here how I am capturing it:
>
> http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
> within-the-dialplan
Or simply use one fo these:
${SIPCHANINFO(peerip)}
${SIPCHANINFO(recvip)}
${SIPCHAN
Hi!
> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?
Use option a of Authenticate together with ${CDR(accountcode)}
Philipp
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> Well, I¹ve tried this, and something just isn¹t right.
Look here:
> Event: Hangup
> Channel: SIP/ShoreTel-1-0004
> Cause: 17
> Cause-txt: User busy
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Hi!
> Can anyone think of a way to play IVR after conversation initiated by
> Dial() terminates?
You will most probably have to prevent the hangup to happen in the first
place:
You could, for example, join the two callers by the help of a dynamic
MeetMe room, and then take action when the oth
Hi!
> A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip
> phones, and we had the opportunity to unregister user by typing *-a
> number and -* again, ex * 99 *, and then the phone number/sip extension
> was unavailable
It is entirely up to you to design the Asterisk dialplan th
Hi!
> but i want to answer the channel when dial someone and pick up the
> phone.not play a file.
Search this list for "early media" and maybe also for "progress".
Philipp
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Hey Gilles,
for whatever reason your messages appear twice twice on this list.
Philipp
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New to Asterisk? Join us for a live introductory webinar every T
Hi!
> But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip.
> No iax. Why does the asterisk machine have to resolve any address?
Probably because you have one or more "register =>" statements in your
sip.conf and Asterisk is trying badly - but without success - to register
Hi!
> exten => **XX
>
> --> This is a local extension, a certain phone which is monitored with
> BLF-lights. So if I press the button I want the phone call that made this
> phone ring, not another phone.
This is NOT a local extension: It is a special local PICKUP extension
(you even named it "[
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