Re: [asterisk-users] test please ignore

On Wed, Nov 10, 2021 at 09:08:52AM +, Kingsley Tart wrote: > my last few emails to this list haven't appeared so I'm just testing 1. Check the archive: http://lists.digium.com/pipermail/asterisk-users/2021-November/thread.html 2. Check your list settings (e.g: Receive your own posts to the

Re: [asterisk-users] clarification on gosub, macros and AEL

On Fri, 2019-10-11 at 14:12 -0400, Brian J. Murrell wrote: > I'm trying to clarify my understand of gosub, macros and AEL. > > My understanding is that macros using the Macro() application, which > is > defined in extensions.conf by: > > [macro-foo] > ... > >

[asterisk-users] clarification on gosub, macros and AEL

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXX.,n,Macro(fastbusy) is deprecated in favour of Gosub().

[asterisk-users] IPv4 address in SDP o= is (null) when configured for NAT using pjsip

Using Asteirsk 13.28.1: If I configure my pjsip transport to handle NAT from the Internet: [transport-tcp] type=transport protocol=tcp bind=10.75.22.8:5060 local_net=10.75.22.0/24 external_media_address=[external address redacted] external_signaling_address=[external address redacted] When a

Re: [asterisk-users] if statement with true value that contains a colon

On Fri, 2019-09-13 at 14:21 +0200, Administrator TOOTAI wrote: > > Escape it with \ Tried that. It doesn't work. Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and

[asterisk-users] if statement with true value that contains a colon

How can I use an IF statement with a true value being a variable that has a colon in it? The colon in the true value variable is being taken as the delimiter for the false value. The only solution I came up with was some hackery to use STRREPLACE to replace the : with a % before the IF statement

Re: [asterisk-users] Preventing rewrite of To: address in MESSAGE transactions

On Tue, 2019-07-16 at 16:20 -0400, Brian J. Murrell wrote: > Is there any option to prevent Asterisk from rewriting the To: > address > of a SIP MESSAGE that it's received and will forward to another SIP > client? > > That is, when Asterisk receives a MESSAGE with the

[asterisk-users] Preventing rewrite of To: address in MESSAGE transactions

Is there any option to prevent Asterisk from rewriting the To: address of a SIP MESSAGE that it's received and will forward to another SIP client? That is, when Asterisk receives a MESSAGE with the To; header saying: To: and wants to forward that to foo@10.75.22.100, I'd like the To: header to

Re: [asterisk-users] RHS of the To: address in MESSAGE transactions

On Thu, 2019-06-06 at 09:33 -0400, Brian J. Murrell wrote: > I'm trying to use linphone-android with asterisk but there is an > aspect > of the way asterisk and linphone-android interact with MESSAGE > transactions that is causing problems. > > The linphone-android f

[asterisk-users] RHS of the To: address in MESSAGE transactions

I'm trying to use linphone-android with asterisk but there is an aspect of the way asterisk and linphone-android interact with MESSAGE transactions that is causing problems. The linphone-android folks consider both the To: and From: address in MESSAGE transactions when deciding which "chat" to

Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

On Wed, 2019-04-17 at 13:50 -0400, Joshua C. Colp wrote: > > The same escaping should apply there for extensions.conf as it's a > config file thing, I don't use AEL and don't know anything in that > regard. It may work the same way there. How very odd. It is working now. I am sure I did

Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote: > On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote: > > > > I can add it onto the end of the variable in the Dial() command: > > > > Dial(${FRED};transport=tcp,${timeout},TtWw); [ the part you trimmed

Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote: > > You specify the transport in the SIP URI. For example: > > sip:t...@example.com;transport=tcp Hrm. This is probably going to be pretty basic, but some googling didn't seem to come up with anything. How do you do this when you are

[asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

Hi, I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport: [transport-tcp-ipv6] type=transport protocol=tcp bind=[2001:1234:5678:abcd::2]:5060 I also have an IPv4 version of that: [transport-tcp-ipv4] type=transport protocol=tcp bind=10.75.22.8:5060 I've then configured an

Re: [asterisk-users] Message: Authentication failed on manager interface

On Thu, 2019-04-04 at 15:08 +0200, Antony Stone wrote: > > It's not "Password", it's "Secret" :) Ha ha. I knew it would be a head-smack type problem. Cheers, b. signature.asc Description: This is a digitally signed message part --

[asterisk-users] Message: Authentication failed on manager interface

I'm not sure how much more simple I can make this but I just cannot seem to get my Asterisk 13 to accept a connection on the manager interface: --- manager.conf --- [general] enabled = yes port = 5038 bindaddr = 127.0.0.1 [myasterisk] secret=a permit=0.0.0.0/0.0.0.0 read = all write = all So,

Re: [asterisk-users] best practices for dialing multiple contacts of multiple extensions

On Thu, 2019-02-21 at 11:17 -0500, Brian J. Murrell wrote: > In the past, I have created variables that hold multiple extensions > such as: > > HOUSEPHONES=PJSIP/mom/grandma > > so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple > phones. > > B

Re: [asterisk-users] pjsip: don't require authentication from remote i register to

On Fri, 2019-03-01 at 15:54 -0500, Joshua C. Colp wrote: > > That's correct. You'd either need to retrieve the line parameter from > the outbound registration or forge the source IP address, Can I eliminate the identify by IP address then, given that my ITSP is supporting the line parameter? Or

Re: [asterisk-users] pjsip: don't require authentication from remote i register to

On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote: > > I don't understand what you mean. Your ITSP has stated that they > don't want you to do authentication with them, so you can't. They are implying, as I am understanding them, that somehow SIP packets they send me shouldn't need to be

Re: [asterisk-users] pjsip: don't require authentication from remote i register to

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > you can try line functionality on the outbound registration which > may or may not work[2] (requires the upstream to adhere to the RFC, > which not all do). My provider seems to implement this. However even with the line=... in the: SIP

Re: [asterisk-users] pjsip: don't require authentication from remote i register to

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > > You either configure IP based matching using an identify section[1] That's what I did: [itsp] type=registration transport=transport-udp outbound_auth=itsp-auth server_uri=sip:pop1.itsp.example.com

[asterisk-users] pjsip: don't require authentication from remote i register to

I'm being told by my ITSP that my Asterisk shouldn't be challenging their system to authenticate (i.e. a 401 response) when they send me a SIP MESSAGE (or I suppose a SIP INVITE for that matter). But I'm not sure what a pjsip.conf configuration for that looks like. How does one associate an

Re: [asterisk-users] PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method

On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell wrote: > I have a PJSIP trunk set up which works fine for voice. I can call > out > and I receive calls from it once it registers. > > What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE) > events. It was wor

[asterisk-users] best practices for dialing multiple contacts of multiple extensions

In the past, I have created variables that hold multiple extensions such as: HOUSEPHONES=PJSIP/mom/grandma so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple phones. But now some of those phones will be registering multiple times and thus have multiple contacts, so I want to

Re: [asterisk-users] branching in extensions.conf?

On Wed, 2019-02-20 at 12:38 -0700, John Kiniston wrote: > I don't see any other messages in this thread other than your initial > one > and my response, perhaps the listserv hasn't relayed it to me yet. I started a new thread:

Re: [asterisk-users] branching in extensions.conf?

On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote: > Use the IF function to evaluate and change the dial command directly. Thanks for taking the time, but that doesn't actually answer the question I asked. It in fact answers the caveat I specifically mentioned: > Granted the particular

[asterisk-users] if function when the true value has a colon in it?

Following up on my previously asked question if I rewrite the branching example (not that it negates the more general branching question) I was using as such: exten => s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})}) exten =>

[asterisk-users] branching in extensions.conf?

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten =>

[asterisk-users] PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method

I have a PJSIP trunk set up which works fine for voice. I can call out and I receive calls from it once it registers. What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE) events. It was working earlier today but I seem to have done something as I was enabling voice on the trunk to

Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote: > When you have an "identify" object configured, you should just use > "ip" as > the "identify_by", But isn't ip the highest priory check in the default value of endpoint_identifier_order and by extension, wouldn't an endpoint without an

Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote: > > What version of Asterisk 13.11.1 I know, I could stand to upgrade. > and what's the value of the "identify_by" > parameter for the endpoint? It doesn't have one. I guess you are implying it should have one. > When you have an

[asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_usern...@sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username

Re: [asterisk-users] (NAT) direct media to host on local net when registering from external address

On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote: > > The chan_sip module has this implemented under the "nat" option using > "comedia" as I recall. Yeah. The help for which reads: Send media to the port Asterisk received it from regardless of where the SDP says to send it. > It causes

Re: [asterisk-users] (NAT) direct media to host on local net when registering from external address

On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > How is your endpoint currently configured in asterisk? It's configured as a chan_sip peer. > Have you tried > rtp_symmetric to see if the endpoint sends audio to asterisk if > asterisk > can send audio back to the client? That would

[asterisk-users] (NAT) direct media to host on local net when registering from external address

This is going to be a bit of an odd situation, but perhaps might become more and more common (as mobile phone SIP clients utilize PUSH proxies instead of the battery draining direct registering with SIP servers). I have a SIP client which can be on the same RFC-1918 LAN as my Asterisk server.

[asterisk-users] messagesend to SIP peer in sip.conf (or otherwise authenticated)

Hi, I want to be able to send SIP SIMPLE messages via/to my VOIP provider but in trying to do so with MessageSend() I am getting 401 errors back from them, unsurprisingly. They want such messages from me authenticated with my account just as they would for SIP voice calls. For voice calls, of

Re: [asterisk-users] CDR_TDS driver disappears - "does not provide a license key" on reload attempt

On Friday 10 Nov 2017, Stefan Viljoen wrote: > Hi All > > I have an Asterisk 1.8.32.3 instance that will at random intervals stop > logging CDR data to MSSQL via FreeTDS. > > On investigation I'll find that the FreeTDS module has been unloaded > somehow. It is not listed in cdr show status or

Re: [asterisk-users] Asterisk Voicemail changes

On Friday 01 Sep 2017, Tim Turpin wrote: > Is there a way that I can modify the source code for the voicemail > application? I need to change some of the options in the user’s interface > to make it work like an existing system that I’m replacing. $ vi /usr/src/asterisk-*/apps/app_voicemail.c

Re: [asterisk-users] Simplest way of executing a non-blocking (async) python AGI script?

On Friday 30 Jun 2017, Jonathan H wrote: > What's the simplest, easiest quickest least-code way of firing off an AGI > with some variable, and then returning to the dialplan? You have to use the "fork" command. This starts a copy of the process with all the same internal state including

Re: [asterisk-users] Autodialer - call simultaneously to both ends

On Monday 26 Jun 2017, Harel wrote: > Hello List, > I'm working on an autodialer project. > At the moment I use the Originate application then I "throw" it to an > extension where I Dial() the other party and then both legs are bridged. > The problem is that the Dial() will only run after the

Re: [asterisk-users] Is this the future of telephony?

On Friday 16 Jun 2017, Christopher van de Sande wrote: > So does anyone here think the traditional telephone company will go > extinct, and voice communication will take place via email like (or > equal to) sip uri's? Hardly! The job of the "traditional telephone company" has always been to

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

On Thursday 15 Jun 2017, Tim S wrote: > Whatever has been done, if anything, isn't working effectively. At this > point I'd like to see some response from the mailing list admin about any > root-cause efforts, AFAIC this is starting to smear the Digium/Asterisk > brand's ability to handle IT

Re: [asterisk-users] Working around missing libmyodbc in Debian Stretch

On Thursday 08 Jun 2017, Olivier wrote: > Hello, > > I'm building a new Asterisk system from source on Debian Stretch. > My building script fails as package libmyodbc is currently missing from > Debian Stretch repo. > > Is there a work around this without leaving MySQL/MariaDB galaxy ? This is

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wednesday 10 May 2017, Steve Edwards wrote: > On Wed, 10 May 2017, J Montoya or A J Stiles wrote: > > Presumably your staff carry mobile phones. What about an app that gets > > the ID of the cell tower to which it is connected, and passes it and the > > SIM number in a HTT

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wednesday 10 May 2017, Steve Edwards wrote: > I have a 'time and attendance' application. Think janitorial or security > kind of thing where an employee goes from location to location. > > They're supposed to 'clock in' when they get to a site using a phone at > that site to prove they're

Re: [asterisk-users] app_jack unavailable

On Wednesday 10 May 2017, andre castro wrote: > Indeed. apt-get install libjack-dev libresample-dev were not installed. > libjack-dev libresample-dev , so I installed. > In the installation of libresample-dev apt-get selected > 'libresample1-dev' instead of 'libresample-dev'. Not sure if that is a

Re: [asterisk-users] app_jack unavailable

On Wednesday 10 May 2017, andre castro wrote: > Hello, > I am new to Asterisk, so please bear with me. > I have made a success installation from source of Asterisk 14.4.0 on > Debian Jessie (8.7). And I am running the Asterisk server, with several > extensions and dialplans, all working well. > >

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017, Frank Vanoni wrote: > By dialing 4000 or 4001, the dialplan is modified and reloaded > accordingly. > > Is there a better solution? That's an . interesting . way of doing things! We would be thinking in terms of using a GLOBAL variable, or an ASTDB entry, to

Re: [asterisk-users] log incoming calls without answering

On Thursday 20 Apr 2017, Fabio Moretti wrote: > Hi, > > I've some analogic lines and I'm asked if it's possible to program an > asterisk for "checking" the inbound calls without answering them, doing > something like this: > > analog line 1 -+-- asterisk > >

Re: [asterisk-users] Voicemail asking for login

On Thursday 20 Apr 2017, D'Arcy Cain wrote: > On 2017-04-20 12:23 PM, D'Arcy Cain wrote: > > Here is the full dialplan for stocktrans2. > > I reduced this to the following and I still have the error. > > exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) > same =>

Re: [asterisk-users] Voicemail asking for login

On Wednesday 19 Apr 2017, D'Arcy Cain wrote: > Yes and [using something like "1571"] works just fine for us. The problem > is that we are trying > to deal with the situation where someone calls themselves from another > phone (internal or external) to pick up their messages. In every other >

Re: [asterisk-users] Voicemail asking for login

On Wednesday 19 Apr 2017, D'Arcy Cain wrote: > On 2017-04-19 02:39 AM, Pete Mundy wrote: > > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > > > > show users' I can't see why the vm_authenticate function is > > failing to read the username :( > > I can answer that one. It's

Re: [asterisk-users] PBX selection

On Monday 17 Apr 2017, Speed Boy wrote: > Hi all, I'm new to VoIP, now we have a project that needs a > PBX with client APPs. > In our team we have argument for choosing PBX. By so far, we > have following candidates: > > A: Open source > > 1) Asterisk PBX (http://www.asterisk.org) (with

Re: [asterisk-users] restart system from extension

On Thursday 06 Apr 2017, Atux Atux wrote: > hi. i would like to be able to reboot the system from my extension. is that > possible? if yes, how? It's possible, with something this in extensions.conf; exten => 99,1,NoOp(Restarting server now) exten => 99,n,System(shutdown -r now) Then

Re: [asterisk-users] Alphabet character in destination number (CDR)

On Thursday 30 Mar 2017, Ikka Tirtawidjaja wrote: > Dear all, > > I have PBX with asterisk 13.x > > a couple of IPPhone that connect to that asterisk PBX send an alphanumeric > dialed phone number. > > for example, in my CDR table, field DST, it show dialed phone number like > - 0C81318304632C

Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

On Saturday 18 Mar 2017, Jonathan H wrote: > Hi, thanks - that looks really good! > > I was about to embark on some non-visual stuff using Ragic, but this > looks great. > > Is there a binary anywhere, or any instructions to compile? I've never > compiled C# code before, and although a quick

Re: [asterisk-users] BUG or ???

On Saturday 25 Feb 2017, Антон Сацкий wrote: > Thanks U Richard > i know about this solution > but the main question why "${} substitution containing > the SHELL is evaluated before anything else" For the same reason why you do raising to powers before multiplications and divisions, and all

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

On Thursday 16 Feb 2017, Max Grobecker wrote: > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client supports TLS, SRTP and ZRTP: > http://phonerlite.de/features_en.htm > > Yes, the GUI is not that much user friendly as Zoiper is - but

Re: [asterisk-users] [OT] Downloading Recqual

On Thursday 16 Feb 2017, Olivier wrote: > Hello, > > While googling, I've just discovered Recqual. > If I'm not mistaken, project's sourceforge site [2] does not host any > source or binary. You need to follow the "code" link, copy the line that starts with "svn checkout ..." and then just

Re: [asterisk-users] Call List Campaign to an IVR

On Thursday 02 Feb 2017, Amelye Chatila wrote: > Hi, > I need to make calls to a list of numbers one at a time and once the user > pick the phone connects to an IVR where I can get few data, after a call > finishes the 2nd number get called and so forth. > > I'm familiar with Asterisk/Elastix

Re: [asterisk-users] Setup DID

On Tuesday 24 Jan 2017, Zakir Mahomedy wrote: > Hi I am trying to setup DDI for one of our servers > Our Provider has given us one DDI for use for eg 080011. > On my main server A, I use an IAX trunk to connect to Client Server > B.When calls come in from the outside world on main server A

Re: [asterisk-users] Replacing PBX during a call in progress

On Thursday 12 Jan 2017, Telium Technical Support wrote: > This was asked many years ago but I thought I would check to see if things > have changed. Is it possible to take over a call in progress - using a > replacement Asterisk server? > > In other words, if 2 user agents are connected through

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

* THIS IS NOT WHERE YOUR REPLY BELONGS * On Tuesday 10 Jan 2017, Olivier wrote: > Historically, I didn't use "install_prereq" but I also used it yesterday. > > As make fails with "[LD] libasteriskpj.o -> libasteriskpj.so.2" which is > the first of its "kind", I still wonder > if issue

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

On Tuesday 10 Jan 2017, Olivier wrote: > Hello, > > I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. > > I followed this: > cd /usr/src > wget ... asterisk-13.13.1.tar.gz > tar zxf asterisk-13.13.1.tar.gz > cd asterisk-13.13.1 > ASTERISK_CONFIGURE="--libdir=/usr/lib64

Re: [asterisk-users] failing to start asterisk on centos7

** THIS IS NOT WHERE YOUR REPLY BELONGS ** On Monday 12 Dec 2016, christopher kamutumwa wrote: > Hello support, > > Am not winning need your help. ive tried putting a different version of > asterisk on centos 7 and here are below results, after make config; > > [root@localhost

Re: [asterisk-users] failing to start asterisk on centos7

On Saturday 10 Dec 2016, christopher kamutumwa wrote: > ive installed asterisk but below is what am getting proces gets > killed.please help > Make sure you have libncurses5 and its development files installed, otherwise this can cause crashes. Also, how much RAM is in your box? Check

Re: [asterisk-users] Asterisk compatibility with SMS services

On Wednesday 30 Nov 2016, Emiliano Vazquez wrote: > i'm using gammu[1] with a 3g dongle and my own chip with my preffer > provider. It can send over 700 every our and receive to. I don't know if > you need asterisk and sms in the same way but with this tool you can make > everything. It has python

Re: [asterisk-users] _FAX_. extension refuses to work !

On Wednesday 30 Nov 2016, Michele Pinassi wrote: >[stuff deleted] > but on a call directed to, es. FAX_3700 i got: > > [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309 > handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension > 'FAX_3700' rejected because extension not

[asterisk-users] Triggering an AGI script when a queued call is answered

Many years ago, I used to have an AGI script that fired on an incoming call, did some database lookups and ended up raising a notification on the screen of the person whose phone was ringing, with the details looked up from the incoming caller ID. All that fell by the wayside when Debian

Re: [asterisk-users] info about DID.

On Thursday 27 Oct 2016, KyD wrote: > Hi! > > I need to make a dialplan by DID. > > where it gets the asterisk values did? from sip headers or ... ? > > Thanks! It will all be taken care of for you, so you don't have to do anything special for calls to a direct inbound number. When a call

Re: [asterisk-users] Incoming Call by DID

On Wednesday 26 Oct 2016, KyD wrote: > Hi, > > My sip provider gave me 2 numbers for the incoming call via pstn. > > nro1 = 12341234 > nro2 = 45674567 > > I have a dialplan for each. > if i put this on my dialplan: > > exten => s,1,Dial(SIP/1001) > exten => Hangup() > > Works! > > But if i

Re: [asterisk-users] Opus codec in codecs.conf

any git tag in particular? (13.12.0-rc1 is my best guess) jrun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016

Re: [asterisk-users] Hello again

On Friday 30 Sep 2016, aaberga/gmail wrote: > Hi, > > after a long pause (Asterisk 1.8 times), I have started again playing with > VOIP. A lot has changed since last time I did setup an Asterisk system! > > So I am asking for some help. [stuff deleted] > [2102] > type=endpoint > context=internal

Re: [asterisk-users] mysql phonebook

On Thursday 15 Sep 2016, tux john wrote: > hi. i am running asterisk 11 and i am using astdb to store all my contacts > and their numbers. so everytime they call me, i can see their name on the > screen of the phone. i am making use of the following to retrieve the name > from the astdb exten => >

Re: [asterisk-users] Multiple phones when one is unregistered

On Tuesday 30 Aug 2016, D'Arcy J.M. Cain wrote: > I have an extension that looks like this: > > exten => 55,1,Verbose(Door buzzer calling) > same => n,Dial(SIP/user1/user2/user3) > > The idea is that any of the three users can answer the phone to let > someone in. The problem is that

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

On Thursday 04 Aug 2016, Nabeel wrote: > On 30 July 2016 at 19:32, D'Arcy J.M. Cain wrote: > > Not playing the prompt changes nothing. If someone presses '*' while > > listening to your answer message then they are in your mailbox. You > > better have a password that they need to

Re: [asterisk-users] SIP trunk

On Tuesday 26 Jul 2016, Jerry Geis wrote: > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have something like: > exten => 55,1,Goto(,yyy,1) > > but if 55 does not come across and it appears to

Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

On Wednesday 20 Jul 2016, Yves biganiro wrote: > Hi all > > Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial > system , Is this some kind of one-stop, pre-prepared distribution with Linux, Asterisk, DAHDI, a web server and some custom scripts, that all installs from

Re: [asterisk-users] VoiceMail and SMS

On Friday 15 Jul 2016, Joaquin Alzola wrote: > Hi Guys > > I am asking too many questions because we would like to use Asterisk first > as a proof of Concept and check from there were it goes. > > - Does the Voicemail have the option of SMS notification on new drop > messages (we have an SMSC so

Re: [asterisk-users] VoiceMail Audio playing

On Friday 15 Jul 2016, Joaquin Alzola wrote: > Hi Madushan > > Maybe I was not clear …. After SIP negotiation and SDP set up on the > VoiceMail Server …. > > Is there a file to specify a MGw (the machine that deliver RTP packages to > end user)? No. The VoiceMail server takes care of all that

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

On Thursday 14 Jul 2016, Joshua Colp wrote: > Carlos Chavez wrote: > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > (language, dtmf, vmexten, etc) and just leave many fields in the > > database as NULL. What would be the proper way to do this for Asterisk > > 13 and

Re: [asterisk-users] Function SHELL not registered

On Wednesday 06 Jul 2016, Michael Jepson wrote: > Adding live_dangerously did the trick. Thanks! But how dangerous is > Asterisk living now ? I must admit, still using an ancient Asterisk version, I didn't know about live_dangerously. But it sort of makes sense. It is somewhat dangerous to

Re: [asterisk-users] rasberry pi

On Wednesday 06 Jul 2016, John Novack wrote: > AstLinux can be remotely managed with the GUI, > which unlike other Asterisk GUI's the conf files are not modified by the > GUI and can be edited "by the book" AstLinux will NOT work with a Pi > though. It is not for the ARM processor. What stops it

Re: [asterisk-users] Function SHELL not registered

On Monday 04 Jul 2016, Michael Jepson wrote: > Hi all, > > I am getting the following error when starting asterisk: > pbx_functions.c: Function SHELL not registered > > Some of my conf files use a SHELL command, which used to work with an older > version of asterisk, but now with version 13.9.1

Re: [asterisk-users] Including doesn't have any effect

On Monday 06 Jun 2016, Markus wrote: > Hi AJ, > Am 06.06.2016 um 10:14 schrieb A J Stiles: > > But why not call an AGI script, have this check the caller ID against a > > MySQL database and return a status -- blocked or not -- in a variable? > > Then you can manage in

Re: [asterisk-users] Including doesn't have any effect

On Saturday 04 Jun 2016, Markus wrote: > Hi list, > > n00b question, but I can't figure it out: > > [callthrough] > exten => _+X.,1,NoOp(nothing here) > #include "blockedall.conf" > exten => _+X.,n(hangup),Hangup > exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" = > "anonymous"]?nocli:cli)

Re: [asterisk-users] Questions... connecting Asterisk to the World

On Saturday 14 May 2016, Stefan Becker wrote: > Greetings, > > asterisk list and community, > > I have a problem in how our telefon switch (Siemens HiCOM) > "talks" with my new configured Asterisk server (V.11.18.0) > > without my Asterisks server in the middle > > <--> Siemens HiCOM

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

On Monday 09 May 2016, Jonathan H wrote: > . {stuff deleted} . > [streamdemo] > exten => s,1,Answer > exten => s,2,BackGround(menu) > exten => s,3,WaitExten > exten => s,4,Goto(s,2) > exten => > _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2)) > exten =>

Re: [asterisk-users] click2call for conferencing two mobile numbers

On Friday 06 May 2016, Alok Srivastava wrote: > Dear List > wanna configure click2call in such a manner that my asterisk box call two > mobile numbers and connect both numbers to talk. I have configured voip > gateway, my internal and external calls are working fine. > please help , You ought to

Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

On Wednesday 04 May 2016, Mamadou NGOM wrote: > Hello everybody, > When I call my extension the agi script don't work well. when I look at > the cli, that is what I have: > [stuff deleted] > AGI Tx >> agi_arg_1: 56 > AGI Tx >> > AGI Rx << SET VARIABLE ** 2 > AGI Tx >> 510 Invalid or

Re: [asterisk-users] my dahdi dont'n start

*** THIS IS NOT WHERE YOUR REPLY BELONGS *** On Friday 29 Apr 2016, Mamadou NGOM wrote: > Hello, > I have not resolved my problem.I renamed my dahdi file "mv dahdi.bash > dahdi " in the directory /etc/init.d, but it doesn'nt work yet. the same > error after the command /etc/init.d/dahdi

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

On Thursday 28 Apr 2016, Robin Kipp wrote: > Hi all, > > sorry if the subject is a bit confusing, but I just couldn’t think of a > good way of better describing the situation… > > Basically, I travel a lot and have several SIM cards for my phone from > local carriers. What I’d like to do now is

Re: [asterisk-users] my dahdi dont'n start

On Thursday 28 Apr 2016, Mamadou NGOM wrote: > Hello, > it doesn't work my dahdi yet .for information, i use debian 8 . > I put the file dahdi.bash in /etc/init.d and I gave it the permission 755 > but i have the same error: bash: /etc/init.d/dahdi: No such file or > directory You need to

Re: [asterisk-users] my dahdi dont'n start

On Tuesday 26 Apr 2016, Mamadou NGOM wrote: > Hello, > > Having installed DAHDI to be able to use the meetme() application , when I > start the dahdi service it generates me the following error: -bash: > /etc/init.d/dahdi: No such file or directory > I need help please. You are using a

Re: [asterisk-users] recreating extensions.conf from live dialplan ?

On Wednesday 13 Apr 2016, Jeremy Kister wrote: > On 4/13/16 11:57 AM, A J Stiles wrote: > > You could try > > *CLI> dialplan show > > Between my older backup and dialplan show, I guess that's my best shot. > > Thanks :D I'll have a go this lunchtime at knocking up

Re: [asterisk-users] recreating extensions.conf from live dialplan ?

On Wednesday 13 Apr 2016, Jeremy Kister wrote: > with the slip of a finger, i destroyed by extensions.conf (grep -i > > extensions.conf) > > I have a backup that is dozens of hours of code old. > > is there a way i can use the asterisk cli (or some other asterisky > method) to recreate that

Re: [asterisk-users] Best timing source?

On Tuesday 05 Apr 2016, Mamadou NGOM wrote: > Hello, > I am doing a configuration for connecting my server asterisk to a SIP > provider. I ask if somebody can give me a basic code or a link to begin > well; Thanks Rule One: Start your own topics -- don't jump in on someone else's, unless

Re: [asterisk-users] Doing asteriksk with a sip trunk

On Thursday 31 Mar 2016, Mamadou NGOM wrote: > Hello ! > I ask if it is necessary to install DAHDI and LIBPRI if we want to connect > our asterisk to an operator SIP (trunk SIP). Someone for helping me. > thanks !!! No. DAHDI is a library for hardware interfaces to POTS, ISDN and mobile lines.

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

On Wednesday 30 Mar 2016, Vitor Mazuco wrote: > Humm thanks for your reply, > > Do you know whats is step for I can transform this card link a fax modem? Start with the specification document for the modulation scheme you want to implement, and the DAHDI Source Code for the card you want to

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

On Wednesday 30 Mar 2016, Vitor Mazuco wrote: > Hi! > > Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or > any others digium card FXO for use Fax modem? Yes, in theory it is entirely possible to use an FXO card driven by software as a modem (and indeed, this is exactly what

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