[asterisk-users] Problem with pjsip

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character

Re: [asterisk-users] Problem with AudioCodes MP-114 ATA

nd restore their config... do you have a backup? regards, yves Am 10.01.2019 um 19:51 schrieb Tech Support: All;     I have an AudioCodes MP-114 four FXS ATA that recently stopped registering to my PBX. I’m pulling my hair out here trying to figure out the root cause without much success. Does a

Re: [asterisk-users] Use AGi Commands without script in Dialplan

Am 08.10.2018 um 13:02 schrieb Antony Stone: On Monday 08 October 2018 at 12:44:43, Yves wrote: I am looking for an easy way to execute any AGI Command directly from the dialplan without the need to call an external script. The whole point of AGI is that it calls an external script in order

Re: [asterisk-users] Use AGi Commands without script in Dialplan

Am 09.10.2018 um 13:56 schrieb Joshua Colp: On Mon, Oct 8, 2018, at 7:44 AM, Yves wrote: Hello, everybody, often it is necessary to issue a single AGI command... How can I realize this within a normal dialplan processing without having to go the circumstantial way through an AGI script every

[asterisk-users] Use AGi Commands without script in Dialplan

a "normal" Call within the dialplan... and again, this is just an example. I am looking for an easy way to execute any AGI Command directly from the dialplan without the need to call an external script. Thank you, Yves -- __

Re: [asterisk-users] Trying to add MoH to conference bridge

could you switch asterisk to verbose >=3 and show the output from the cli? which version of asterisk do you use? yves Am 23.05.2018 um 23:23 schrieb Mike Diehl: Hi all, I've got an AGI script that launches the conference bridge with a line like: "$main::agi->exec(ConfB

Re: [asterisk-users] Looking for better fax handling

of course you can query asterisk asterisk and look, if your fax is still running...: asterisk -rx "fax show sessions" lists you all acive fax sessions... yves Am 22.05.2018 um 12:19 schrieb D'Arcy Cain: On 2018-05-22 02:17 AM, Yves wrote: you could - use "global v

Re: [asterisk-users] Looking for better fax handling

you could - use "global variables" - use the asterisk built in database - mv the file to temporary folder _before_ faxing (would be the most easy solution as you already know how to mv a file via asterisk...) regards, yves Am 21.05.2018 um 19:49 schrieb D'Arcy Cain: I am havin

Re: [asterisk-users] pcapsipdump or general sip debug question - the solution

merge" 3.) go to "telephony -> sip flows" 4.) select the two "legs" of the call 5.) klick button "flow sequence" et voilà... one ladder diagram exactly the way I needed it thanks anyways, yves Am 17.01.2017 um 12:34 schrieb Jean Aunis: Hello, There i

[asterisk-users] pcapsipdump or general sip debug question

the full sip flow between both ends of one call in a single file (per call) with pcap compatibility (including the rtp packets)? thank you yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: On Wed, Dec 21, 2016 at 7:50 AM, Yves <yves...@gmx.de> wrote: Hi Mark, yes, you are right... these are different VLANs I configured the other

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

? Which software-versions are you using? thank you, yves if someone wants to take a look at the phone-logs: boot-log 02.335|so |*|01|-- Initial log entry -- 02.335|so |*|01|+++ Note that Updater log times are in GMT +++ 02.335|boot |*|01|Initial log entry. Current

[asterisk-users] Polycom SoundStation IP 6000 does not register

Hi, I am pulling my hair for days now... I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with my Asterisk. There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... Simple Question: Does anybody have a

Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

ok, thank you... then I´ll take it as it is cheers, yves Am 18.12.2016 um 13:15 schrieb Larry Moore: Hi, I haven't found anything definitive however I expect the TSI that is sent during initial fax call establishment is stored by the receiving terminal, see pages 28 & 29 of the Eng

Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

e settings like font, position, and so on? thanks, yves Am 18.12.2016 um 00:02 schrieb Larry Moore: The list of options available are listed here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT It doesn't appear that a received header is available unless it is written

[asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

to only exist the Faxopts remotestationid but for sure on any fax I receive there is a remoteheaderinfo besides the remotestationid... it is on the tiff-file, but I need this info in a channel-variable... Does anybody know how to get the remoteheaderinfo for a received fax? thanks yves

[asterisk-users] WhatsApp feature on Asterisk

Can anyone put light on whatsapp features and how it can be operated . What are the technology that need to be installed , Regards Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

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

Asterisk 1.8.23.0-1_centos5.go DAHDI Version: 2.6.1 Echo Canceller: HWEC On Wed, Jul 20, 2016 at 5:32 PM, A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > On Wednesday 20 Jul 2016, Yves biganiro wrote: > > Hi all > > > > Hi,I'm facing a strange issue wh

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

I have forcefully installed everything but it says that the card is not found. On Wed, Jul 20, 2016 at 5:05 PM, Yves biganiro <yves.bigan...@gmail.com> wrote: > Hi all > > Hi,I'm facing a strange issue where by SANGOMA not detected by > goautodial system , Thats the proble

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

Hi all Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial system , Thats the problem : Configuring ISDN BRI cards [A500/B700] No Sangoma ISDN BRI cards detected Press any key to continue:

[asterisk-users] open source pbx free

Anyone have any experience running an open source pbx and call center solution?Need to start a call center of 10 users and i need help I have already installer a server with Ubuntu Server 14.04 , E1 installed Please advice me how to process from here Regards Yves

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

Hello, I was faced with this problem, it is enough to place subdirectory under ./tools installation dahdi when compiling and run make install-config it should work. we must have : mkdir -p / etc / dahdi mkdir -p /etc/modprobe.d install -m644 xpp / genconf_parameters / etc / dahdi /

Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

Le 06/04/2016 18:12, Markos Vakondios a écrit : Good evening, My English is limited but if I can help. We install Asterisk Version 13.1 on VmWare with Debian 8.2, no problem since June 2015, currently I have tested on Unbutu 14.04 but problem with network-manager (problem of stability with

Re: [asterisk-users] The To header was truncated in call... Whats this means?

I have seen these messages only on asterisk boxes that are open to public and I think this may have something to do with sip-attacks... I´d recommend some wiresharking or at least sip debugging... yves Am 07.01.2016 um 21:23 schrieb Vitor Mazuco: Hi everybody, My Asterisk, all time appear

[asterisk-users] placing calls with linphone.org SIP account

ng looking like? (I am trying with zoiper softphone) Unfortunately there is no support-email-address for linphone.org users... thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com

[asterisk-users] No QueueCallerJoin Event...

to an agent... Is it a bug or am I missing something? regards, Yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] asterisk-java is dead?

No, its not dead and mails to the asterisk-java-list become replied. regards, yves Am 18.09.2015 um 02:35 schrieb symack: Hello Everyone, I am trying to make use of asterisk-java live and had some questions for the mailing list however, it does not seem like it's an active mailing list

[asterisk-users] call failed... but why? What means SIP_ALREADYGONE?

? thanks for watching, yves SIP Phone 110 (callerid 061444018110) tried to call the external Phone Number 0616677823 and gets an hangup after 2 seconds. Another try immediately after the failed call goes fine. The failed call did not arrive at the destination. [Feb 12 10:00:11] DEBUG[1567][C

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

Hi, I know this Bug,,, at least when you´re talking about x-lite 3... quite annoying, but if you know it... so no... its not the phone... tested with zoiper and 3cx ... both work...but the problem occurs ONLY, as soon as I register at more than one registrar... yves Am 22.11.2014 um 19:19

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

Hi, the useragents nothing to do with the problem... i tried numeric, alpha and alphanumeric... no difference. they work all as long as I only use ONE registrar... as soon as I register at more than one registrar... the line drops after 32 seconds really strange. yves Am 22.11.2014

[asterisk-users] how to set timerb in sip.conf

for timerB. How can I configure the timerb value? thx, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Softphone signals busy although it isn´t

-realtime peers with mysql. thanks for reading, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] SIP call drops after 32 seconds, but only when....

32000ms, and it does not make any difference if I configure timerb in the general context or in the phone context... any ideas? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

[asterisk-users] Best strategy to find and solve voice quality problems

QoS (google only showed me such settings for Lync or Windows Server machines...? Is there a way under Windows XP / 7 to ensure CPU-Bandwidth for Applications (like VoIP Clients)? Thanks for any hint, yves -- _ -- Bandwidth

Re: [asterisk-users] how to send dtmf after pause ?

This would be possible with an agi... the agi can wait for silence or 10 seconds, as u like and then play the dtmf tones and bridge the call to your extension afterwards. yves Am 07.06.2013 17:51, schrieb Sean Darcy: I'm trying to call a conference service, wait 10 seconds, then send

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

looks yummy indeed... but how does it interact with an asterisk? phono uses afaiu voxeo-cloud to make place calls, send sms and so on... I do not see a way to use phono without their cloud services, not did I see any hint about charges for calls... yves Am 03.06.2013 12:34, schrieb Lenz

Re: [asterisk-users] Help me understand these log messages

... an anonyous (not registerted) sip user from 188.161.238.232 was trying to initiate a call to 9725955 and so on... you could enable sip tracing to get more information. maybe you should change the 'allowguest' option in sip.conf..? regards, yves Am 31.05.2013 23:57, schrieb Chris Gentle

Re: [asterisk-users] Executing a dynamic sequence of applications

Hi, I would recommend an AGI-script or a realtime dialplan for this purpose. yves Am 30.05.2013 11:46, schrieb Grant Bagdasarian: Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications

Re: [asterisk-users] Sangoma Wanpipe Driver

- solved - it turned out that libpri was not compiled correctly... and... Asgars comment about group systax is correct. thx regards, yves Am 13.05.2013 13:21, schrieb Yves A.: that was the syntax before 1.8 or 11.x I think... what about pseudo? yves Am 13.05.2013 13:16, schrieb Asghar

Re: [asterisk-users] amiDebugger - might make your life easier if you program through the AMI

thank you! such efforts for the community are always highly appreciated! - I´ll give it a try. regards, yves Am 13.05.2013 21:44, schrieb Lenz Emilitri: Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story

[asterisk-users] Sangoma Wanpipe Driver

successfully compiled sangoma driver 7.0.1 in combination with an asterisk 11.3? thanks for hints, regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Sangoma Wanpipe Driver

mmh... actually supportline is closed... why proceeds the call to dahdi/pseudo-?? i have never seen this before... thx., yves Am 13.05.2013 11:42, schrieb Duncan Turnbull: We have had challenges with the latest kernel versions on Ubuntu and sangoma wanpipe drivers An older kernel

Re: [asterisk-users] Sangoma Wanpipe Driver

that was the syntax before 1.8 or 11.x I think... what about pseudo? yves Am 13.05.2013 13:16, schrieb Asghar Mohammad: Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number? On Mon, May 13, 2013 at 12:53 PM, Yves

Re: [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1

hi, i would try to make a symlink... link the wrong folder to the correct one... yves Am 02.05.2013 23:34, schrieb James Mortensen: Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org

Re: [asterisk-users] looking for a way to do appointment reminders

installation without any issues. regards, yves Am 26.04.2013 03:55, schrieb Brandon Coale: Hello, My health care organization is looking for a way to do appointment reminders. We currently have staff members who spend part of each day manually calling patients to remind them of their upcoming

[asterisk-users] PRI DEBUG

debugging? thx, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users

Re: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes

Hi, I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and would say it is a bug... To remotely hang up a call use * **hangup request channel* where channel is the exact id of your channel as you would receive it via *core show channels* yves Am 11.04.2013 10:56, schrieb

Re: [asterisk-users] PRI DEBUG

thanks, that command syntax works. yves Am 11.04.2013 18:51, schrieb Richard Mudgett: - Original Message - hi, strange behaviour while trying to use pri debugging on asterisk 11.x ... please take a look: bas1104*CLI pri show version libpri version: 1.4.13 bas1104*CLI dahdi show

Re: [asterisk-users] dahdi-channels.conf vs. chan_dahdi.conf

or look in you /etc/dahdi/modules if you disabled the loading of the module for your newly added card. after this run dahdi_genconf and all should be set up atomagically... regards, yves Am 28.03.2013 14:44, schrieb Ken D'Ambrosio: Hey, all. Just added an analog card to our dual-T1 system

Re: [asterisk-users] WARNING[28151] from CLI

, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] WARNING[28151] from CLI

sometimes, but for me, thats all i can tell about. regards, yves Am 27.03.2013 13:06, schrieb Salaheddine Elharit: thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com mailto:t...@ovm

Re: [asterisk-users] question about zapata.conf

it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44

Re: [asterisk-users] question about zapata.conf

it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application

Re: [asterisk-users] Recording with MixMonitor and AGI

hi, the music heard by MoH is configurable... so if you want silence... But hold could e.g. also be done by transferring a caller into a dynamic meetme room... yves Am 14.03.2013 08:43, schrieb Henrik Westerberg: Hi, The idea was to record an ongoing call by three party bridging

Re: [asterisk-users] Recording with MixMonitor and AGI

are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1

Re: [asterisk-users] Sending SMS from asterisk

...) if it would be possible to use the SMS ServiceNumber from my mobile Provider...? I have a valid mobile contract, the number of the SMScc , my Cardnumber (t-mobile), my phonenumber and so on... so it should be possible, I think... but how? Has anybody a clue? regards, yves Am 09.03.2013 11:03

Re: [asterisk-users] Recording with MixMonitor and AGI

to be taken care of and depending on your agi... it might be interrupted, if the call is hungup... but as you did not show your agi... these are just hints.. regards, yves Am 07.03.2013 16:21, schrieb Henrik Westerberg: Hi, I am developing a call recording application on Asterisk 11.2 and have

Re: [asterisk-users] Extension cant pickup calls but can transfer.

do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions

Re: [asterisk-users] Extension cant pickup calls but can transfer.

is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who

Re: [asterisk-users] Extension cant pickup calls but can transfer.

... check cli output and sip debug output... good luck. yves Am 07.03.2013 20:38, schrieb Luis H. Forchesatto: Yes, both are configured in the same ata (linksys pap2) and the configuration options are the same. Call group and pick group are the same for both too. 2013/3/7 Yves A. yves...@gmx.de

Re: [asterisk-users] red alarm on span - do channels in the group automatically get skipped over?

hi, yes, this is the way, asterisk / the channeldriver handles it. you can simulate the failure of one span by just pulling out the cable and see what happens.. on top, you can influence the order, the channels are used by using dahdi/g1 or dahdi/G1... regards, yves Am 05.03.2013 07:31

Re: [asterisk-users] ODBC and SQLIte3

to caching or other reasons), the ; information will not be removed from realtime storage regards, yves Am 17.02.2013 12:51, schrieb termo termosel: Hi, I had configured Asterisk to use default database located in /var/lib/asterisk/sqlite3dir/sqlite3.db. When I

Re: [asterisk-users] ODBC and SQLIte3

looks like a mistake in your extconfig.conf... do you want to use realtime extensions too? for further instructions show us your extensions.conf and the verbose output of the cli showing the dialattempt... regards, yves Am 17.02.2013 14:31, schrieb termo termosel: Hi, I have add

Re: [asterisk-users] cisco 7940 and asterisk 11

... if all fails, i would then go deeper into network analysis and trace the traffic. meanwhile i administer around 10 asterisk boxes and i always use ubuntu 12.04 lts and latest asterisk 11 on dell r3/4/6xx servers... up to now everything runs fine.. regards, yves Am 14.02.2013 07:20, schrieb

Re: [asterisk-users] Variables set by AGI lost in dialplan

could try to set the variables with help of the shared function to set it for both channels... regards, yves Am 14.02.2013 10:40, schrieb Deepesh D: Hello, I am using asterisk 1.8.17.0 with a fast agi written in C The following is a part of my dialplan exten = _X.,n,MSet(my_var=0,my_var1=0

[asterisk-users] Asterisk Realtime Extension... strange behaviour

to... as I said.. using static extension via extensions.conf the dialplan works as expected... Am I missing something? regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad

do not work as they have different links) 2.) configure one end as master (CPN) and the other asterisk as Network (CPN), otherwise you´ll get timing issues... thats all... regards, yves Am 11.02.2013 14:00, schrieb Shitian Long: Hello, I am trying to connect two asterisks with PRI connection

Re: [asterisk-users] access control softphone registration through asterisk

... there are hundreds of possibilites... the easiest way I think would be to use the asterisk build-in database (therefore the hint to the function db...) regards, yves Am 08.02.2013 22:18, schrieb Muhammad: Hi, I wana control my SIP register from asterisk. I other hand, when users login

Re: [asterisk-users] special conference room

hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: Hi list

Re: [asterisk-users] special conference room

, that the caller can talk to the moderator or not... any caller should NEVER hear what other callers are talking... may he be muted or not... yves Am 16.01.2013 23:01, schrieb Danny Nicholas: From what I read, neither confbridge or meetme have the whisper feature built-in; This doesn't matter

[asterisk-users] special conference room

- the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves

[asterisk-users] Asterisk console suddenly extremely verbose...

Hi there. I started the console today to reload the extensions.conf file ; only to be greeted with extremely verbose console. Seems related to the zaptel card: Example: Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020

Re: [asterisk-users] Asterisk console suddenly extremely verbose...

Hi On 16 December 2011 13:24, Richard Mudgett rmudg...@digium.com wrote: You have pri intense debug span x enabled. Disable with pri no debug span x. Thanks... I couldn't find any configuration file showing this ; but ran the command in the CLI... Seems to have done it. I really wonder how

[asterisk-users] Redirecting a Channel more than three times...

the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves

Re: [asterisk-users] Redirecting a Channel more than three times...

the dialplan is very complicated, but it showed me no hint of beeing responsible for this... the cli-output gives no hint. yves Am 24.09.2010 15:10, schrieb Danny Nicholas: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio

asterisks´ side, i penetrated the support until they looked over it again and... what should i say... finally they had to admit, that the router had a wrong acesslist. they corrected it and now it works. yves Brian schrieb: On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote: Hi, I am

[asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio

from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? thanks, yves

Re: [asterisk-users] Odd error mssage on DAHDI lines

hi, you can lookup the causes in the sources check you dahdi-configuration (especially the groups...) is there everything ok? what does dahdi_tools or the other cli-commands say, that give you information about the available channels? yves /* Causes for disconnection (from Q.931

Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops

do you use the qualify=yes option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing

[asterisk-users] sip.conf with versatel and two NICs very strange problem

to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves

Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem

thanks, i tried this already but unfortunately no change. any further suggestions or answers concerning my other questions? thanx, yves Cary Fitch schrieb: As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite

Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem

thanx... a typo... the routers local ip is 10.26.208.253 yves Tim Nelson schrieb: - Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has

[asterisk-users] DTMF reception during WaitForSilence

AND detecting DTMF-Input AND detecting silence to know, when Input has finished... (I want to avoid that users have to finish their input with the pound-key...) ? Btw.: why are the DTMF-Tones, that a user enters, not hearable in the recording? Thanks for your help and hints, Yves

[asterisk-users] Zaptel - DAHDI: now echo

=incoming faxdetect=incoming group=1 channel=1-10 --- From reading the various documentation, I was convinced that moving from zaptel to dahdi was almost just a matter of renaming the configuration file... Am I mistaken ? Thank you in advance for any help. Jean-Yves

Re: [asterisk-users] Zaptel - DAHDI: now echo

that there are particular software configuration available , but I haven't had a clue on what they are for, nor did I find documentation about it... I'm not building asterisk nor dahdi myself, but instead rely on packaged from ATrpms.conf Thank you Jean-Yves

Re: [asterisk-users] Zaptel - DAHDI: now echo

Hi 2009/8/18 Tzafrir Cohen tzafrir.co...@xorcom.com: Something is missing here... http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules Thanks .. I added to /etc/dahdi/system.conf the following: echocanceller=mg2,1-10 However, I have no clue about the various echo canceller,

Re: [asterisk-users] Read Command

You first use the Read application : exten = s,n,Read(ANS|filetoplay) And then use GotoIfs by checking the ${ANS} variable to do the logic (re-ask if bad response, else continue in dialplan). On Sun, 2008-08-24 at 23:10 -0700, Joe Carroll wrote: I’ve search the world over…. but I haven’t

Re: [asterisk-users] IVR question

You could use func_odbc in your dialplan, check here : http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc Yves. On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote: Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed

Re: [asterisk-users] IVR question

Sorry, maybe I misunderstood your question. If you want the dialplan to be in a MySQL dabtase, check here : http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database Works great, but the documentation is sometimes a bit outdated. Good luck. Yves. On Thu, 2008-08-21 at 14:57

[asterisk-users] Playback don't play the beginning if a sound file

Hello, I'm using this dialplan to let user record messages. The recording part works quite fine, but there is something strange : When Asterisk plays vm-torerecord, it misses the beginning, I only hear the few last seconds (vm-torerecord is a sound file that was in the asterisk-sounds cvs repo,

Re: [asterisk-users] Playback don't play the beginning if a sound file

It seems this has something to do with the Wait() before the Playback (Background behaves the same). If I remove the Wait, the next Playback is just fine, otherwise it truncates the beginning of the message. On Mon, 2008-05-05 at 10:41 +0200, Yves Räber wrote: Hello, I'm using this dialplan

Re: [asterisk-users] Goto in Realtime extensions

That's very unfortunate. I use now a workaround : I'm just switching (with gotos) between extensions and use some macros but always within the same context. I'll try to remeber it for next time :) Cheers, Yves. On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote: On 2/8/08, Yves Räber

[asterisk-users] Goto in Realtime extensions

? It seems that it's not possible to Goto to another context within the realtime extensions. Cheers, Yves. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Goto in Realtime extensions

I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime

Re: [asterisk-users] Goto in Realtime extensions

that has to generate the contexts). * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED

Re: [asterisk-users] Goto in Realtime extensions

I'm not using labels at all (but I've also tried with :)) On Thu, 2008-02-07 at 16:39 -0800, Grey Man wrote: Make sure you don't have any labels on the prioritys. When loading extensions from realtime labels aren't supported. Replace: exten = _X.,1(mylabel),... with exten =

Re: [asterisk-users] How to prevent logging of some entries in CDR

Hi On Jan 21, 2008 11:05 PM, Jean-Yves Avenard [EMAIL PROTECTED] wrote: This works great. However in the CDR, than seeing one entry for each call, I see several entries in the CDR Worse, if I do something like: Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) 40

Re: [asterisk-users] How to prevent logging of some entries in CDR

Hi On Jan 25, 2008 4:58 AM, John Faubion [EMAIL PROTECTED] wrote: I have the same issue but I haven't put much effort into solving it yet. Too many other issues seem to get in the way. If you do, please post your results ! ___ -- Bandwidth and

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