Re: [asterisk-users] Mobile Phones and Asterisk

e the busy status of a device? > Have you done something similar to this? > > I'm using ver. 1.6. Thanks in advance. I'm not sure I understand your setup. Are you using SIP for trunking, or for extensions? Are you calling a normal mobile phone, or

Re: [asterisk-users] Mobile Phones and Asterisk

line is not ringing out? Is that what is wrong? And why do you have some "xxx" in front of ${extension}? You shouldn't need them. Just pass ${extension} - which is the number you dialled on the phone. Sebastian > > I hear a busy tone, after the 10 sec. timeout it returns

Re: [asterisk-users] Mobile Phones and Asterisk

the sound coming down the line). You might also want to check your regional settings in Asterisk. Sebastian I achieved this successfully by emulating it via a softphone, when I call a softphone and it is currently engaged in a call, asterisk returns BUSY in DIALSTATUS and will automatically fallback t

Re: [asterisk-users] Mobile Phones and Asterisk

ch screws things up - as Asterisk can't tell that the mobile is not available. To Asterisk, that message is the same as somebody answering the line. Same in France and Spain - as far as I've seen. Sebastian > - still no answer that pots line is hung up and call drops back into the &g

Re: [asterisk-users] Asterisk 1.8 Multiple Parking Lots

> // > /exten => 2011,hint,park:2...@parked/ > /exten => 2011,1,Wait(1)/ > /exten => 2011,2,Set(CHANNEL(parkinglot)=parkinglot_A/ Is it just a typo that fact that you are missing a closing bracket on the above line (and two more like it below)? Sebastian > /exten

[asterisk-users] Random call drops on IAX2

1512541010, 3) exited non-zero on 'SIP/23-0089' [Nov 10 11:27:43] VERBOSE[3242] chan_sip.c: -- Registered SIP '23' at 192.168.12.16 port 5064 Any ideas or comments on this one are much appreciated. I'm not sure where else to look to get more relevant informatio

Re: [asterisk-users] Random call drops on IAX2

anybody? On 11/10/2010 06:51 PM, Sebastian wrote: > Hello list, > > I have an Asterisk setup with the following details: > > 1. 3 x internal extensions / sip hardphones - Grandstream 2000 > 2. 2 x internal extensions / dahdi cordless phone > 3. 1 x 2 FSX ports OpenVOX pci c

Re: [asterisk-users] Asterisk Playback sound dropping on linphone

ive to linphone (which seemed good enough anyway!)... I use linphonec as well - and haven't found another console sip phone either. I'd be interested if there is another one. Sebastian > > Thank you, > Matteo > -- __

[asterisk-users] IAX2 and INVAL packets

efore I hung them up) when I tried. So not easy to replicate. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] IAX2 and INVAL packets

On 11/18/2010 10:38 PM, Tilghman Lesher wrote: > On Thursday 18 November 2010 14:01:49 Sebastian wrote: >> > Is anybody here familiar with the meaning of INVAL packets for IAX2? >> > >> > Every few days I get a dropped outgoing call in the middle of the >>

Re: [asterisk-users] IAX2 and INVAL packets

On 11/18/2010 08:01 PM, Sebastian wrote: > Is anybody here familiar with the meaning of INVAL packets for IAX2? > > Every few days I get a dropped outgoing call in the middle of the > conversation (the outgoing call has been connected for few minutes) when > an incoming call co

Re: [asterisk-users] Moving asterisk from one network to another.

, they might contain the IP's of external and internal networks in the config files. Your firewall (if you have one) might contain IP's and network masks. It depends on how the box was originally setup. Sebastian It is assumed that DB is on the same box. Asterisk box has got Asteri

[asterisk-users] Detect missed call in extensions?

to a missedcall.txt log file. (call should be logged in 3 case, but not in 1 case) 2 is easy to detect, as these always are failed (non-answered) calls. Best regards, Sebastian Nielsen smime.p7s Description: S/MIME Cryptographic Signature

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

I would suspect that the cell phone does use battery saving causing the SIP application to lose registration with the server. Would also suggest using TCP with a fairly short keepalive to prevent the cellular network from tearing down the connection to the asterisk server. You need to go into a

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

of ”Registred” to your trunk operator. Från: Ivan Demkovitch Skickat: den 15 november 2018 18:01 Till: Sebastian Nielsen ; 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some reason Sebastian,

[asterisk-users] Need for more hangup reasons in ARI?

those five hangup reasons? Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start

Re: [asterisk-users] Need for more hangup reasons in ARI?

TW: Is there a way to have them documented on the Wiki page instead of having to dig into the source code? I'd be happy to help. Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Paging systems?

How did the page system answer the call when it was used with the analog system? You could propably ”fake” those signals from inside asterisk, and cause it to answer. Från: asterisk-users För Michael Munger Skickat: den 21 mars 2019 20:00 Till: asterisk-users@lists.digium.com Ämne: [asterisk-

Re: [asterisk-users] Paging systems?

Small Business Specialist Digium Certified Asterisk Professional High Powered Help, Inc. p: 678-905-8569 w: <https://hph.io> hph.io e: <mailto:m...@hph.io> m...@hph.io On 3/21/19 3:01 PM, Sebastian Nielsen wrote: How did the page system answer the call when it was use

[asterisk-users] Problems with calls dropping on Android.

Hello. I have the following in sip.conf [sip09] type=peer defaultuser=sip09 nat=yes qualify=no secret=sip09 host=dynamic context=outgoing dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h263p deny=0.0.0.0/0.0.0.0 permit=192.168.2.2/255.255.255.255 jbenable = yes jbforce

[asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

, is it possible to tell Asterisk to just ignore the lack of acknowledgement from Android somehow? Basically, for Client sip09 (username), never hang up for the reason 18 (NO_USER_RESPONSE), threat like user response was received always. Best regard

Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

16, 2019 at 7:45 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote: Hello. I have a problem with the native Android SIP client, not acknowledging the call. Sent a message to the list for some weeks ago containing a sip debug log, but it only got stuck in moderation queue due

Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer

, Nov 16, 2019 at 7:59 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote: What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones? I do not know if the bug is in Android native S

Re: [asterisk-users] On Register, run a script, validate source IP

You could use permit/deny in the sip.conf. That would require your script to update sip.conf dynamically and reload the config for each time user wants to update their accepted location. To avoid excessive reloads, you could have that the changes will take effect after 00:00, so you have a cron

Re: [asterisk-users] Stir-Shaken for asterisk

Yes, this means that a provider which only provides IP-access (for example a broadband operator), ergo, when it doesn’t terminate a call, but where the call terminates directly at a enterprise, does not need to force the end customer to implement call verification in their PBX. Basically, if you

Re: [asterisk-users] Channels freeze on Confbridge

tures, but it turned out it outright rejects packets with unsupported features. Best regards, Sebastian Nielsen -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com För C.Maj Skickat: den 22 augusti 2020 20:03 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-us

Re: [asterisk-users] Channels freeze on Confbridge

>>I can see the point you're making here, but what's going to do this after 30 *minutes* of normal call? I was more into, if there is some feature that somehow triggers after 30 minutes of call - and this feature is unsupported on some client, which causes it to drop the call. For example, if y

[asterisk-users] Inband DTMF not detected - bug or config error?

core bridge, so DTMF tones get detected inband and converted to rtp events? Any hint appreciated. Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk commu

[asterisk-users] Anyone that know of DECT "client" for asterisk?

s very high echoes in the phones. The idea is to have something simulate a DECT handset, connect to the provider's router, and thus be able to still use asterisk. Best regards, Sebastian Nielsen smime.p7s Description: S/MIME Cryptographic Signature -- ___

Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?

ium.com För Frank Vanoni Skickat: den 7 oktober 2020 19:17 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Anyone that know of DECT "client" for asterisk? On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP

Re: [asterisk-users] Detect if people is talking

It sounds like there is more of the problem that neither the agent or customer knows when to start talking, ergo, when the call is "Connected", thus the OP wants the agent to start talking before the customer is brought in front of that agent. Another solution would be to just play a "fake" rec

Re: [asterisk-users] STIR/SHAKEN

company 2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from company 1 to company 2 – then company 2 owns your DIDs. Best regards, Sebastian Nielsen Från: asterisk-users-boun...@lists.digium.com För Alexander Perkins Skickat: den 12 mars 2021 01:23 Till

Re: [asterisk-users] STIR/SHAKEN

carrier, can the DID provider also give you outbound calling? Most likely, but that doesn't mean that the best way to go is to route outbound calls via the carrier that is providing you DIDs. On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote: I rea

Re: [asterisk-users] STIR/SHAKEN

”tre” ( https://www.tre.se ) has in sweden, one of sweden’s largest phone operators, they are 4th the largest phone operator. (1: Telia, 2: Tele2, 3: Telenor, 4: Tre) Then you understand why I wonder WTF people are doing… Best regards, Sebastian Nielsen Från: asterisk-users-boun

[asterisk-users] DECT client adapter

nto a DECT base station, that works with Asterisk? Best regards, Sebastian Nielsen smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out th

[Asterisk-Users] Ring when leaveing queue?

their call is about to be answered. Any ideas? Regards, Sebastian. -- Sebastian Filzek Teragen International Pty Ltd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP agent logging into queue?

-5207) and therefore does not log out. Does anyone know what the data tacked on the end of the SIP name is and how to stop it? Regards, Sab. -- Sebastian Filzek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Can't dial out from SIP to CAPI

ce? I have not tried the callcentric stuff so far and that's not so important for me right now. I just want to be able to dial out in the first step. Maybe you find my problem. Thanks, Sebastian. ___ -- Bandwidth and Colocation Provided by http://www.api-

[asterisk-users] Asterisk bridging timeout when calling out with SIP phone

pears, when I'm calling out from my SIP phone to PSTN. When somebody is calling in from PSTN to my SIP phone the ISDN<->SIP is instantly bridged. Is there any option in SIP.conf oder capi.conf I have to set? Thanks, Sebastian. ___ -- Bandwidt

[asterisk-users] constantly increasing load in Asterisk 11.14

11.14 could cause this behaviour? It looks like we have to go back to 11.6. Best Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory we

Re: [asterisk-users] constantly increasing load in Asterisk 11.14

en was restarted. In November, we updated to 11.14, and from that time, it looks a bit different (and Asterisk needed a lot more restarts). Best Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digit

[asterisk-users] [PoE] Avaya 1152a1x

monitoring of power output using snmpwalk on the device. Does anyone has experience wit such an device? Does anyone know what my fault might be? How I can configure the device to output more information via SNMP? Thank you very much! Sebastian

[asterisk-users] Asterisk on OpenWrt (first time user)

6.64.162.35 [acl_ekiga_outbound] deny=0.0.0.0/0.0.0.0 Only load necessary modules: [modules] autoload=no load => chan_sip.so load => res_rtp_asterisk.so load => app_dial.so load => pbx_config.so load => app_cdr.so load

Re: [asterisk-users] Update peer IP address

st. That should tidy up the dialplan. What I'm a little afraid of is the SIP provider using IPs out of a range that they also use for other services. Maybe out of the same range they hand out IPs to their customers. I guess we got to be careful :-) Kind regards, Sebastian > The Asterisk i

Re: [asterisk-users] Update peer IP address

On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote: > Hello Sebastian, > > I had already seen this list of the hosts, but it is not active. All > servers with which my Asterisk has been communicated are not listed. > > A port scan, to eventually update the list,

Re: [asterisk-users] Update peer IP address

every seconds ; default is 300 (5 > minutes) Hello Andres, I read that same suggestion elsewhere in connection with Deutsche Telekom, so it seems there's some benefit in it. Daniel, did you try it out already? Kind regards, Sebastian -- _

Re: [asterisk-users] Update peer IP address

#x27;s because I'm "hiding" that I'm using Asterisk (sdpsession, useragent, also custom systemname in asterisk.conf). But probably that's not the reason. Anyway, I'm just going to wait until it doesn't work and then worry about it. Regards, Sebastian -- __

Re: [asterisk-users] Update peer IP address

On Tue, Apr 14, 2015 at 09:38:22AM +0200, Daniel Heckl wrote: > Sebastian, > > Your code sounds good, I'm curious how it goes on. > > First the linux machine had the Google Public DNS 8.8.8.8 as DNS > server. After I changed it to the via PPPoE assigned DNS servers, i &g

Re: [asterisk-users] Debugging dialplan

"exim -bt", if someone here know the SMTP-daemon Exim... > > Is there such an option in Asterisk? > Hi Luca, try 'dialplan show @'. Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by h

Re: [asterisk-users] Debugging dialplan

read up on that using the Internet (there are e.g. wiki articles about this subject) or a book (e.g. "Definitive Guide on Asterisk"). Regards, Sebastian Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello : >Zitat von jg : > >> Yes, it is called "core set verbose

Re: [asterisk-users] Signaling incoming call

I see this page (in German): > >http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72 > >Could you please help me? &g

Re: [asterisk-users] Logging in "local time"

>I didn't found in logger.conf or other file an option to set the >timezone. >Can someone help me? > >Thanks >Luca Bertoncello >(lucab...@lucabert.de) Hi Luca, set up a proper /etc/timezone, see http://w

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis : >alsa_card_init^[[0m: snd_pcm_open failed: Connection refused >soundcard_init^[[0m: Problem opening alsa capture device > >These are the errors I get. > >I changed the following: >chown -R myuser:myuser /var/log/asterisk >chown -R myuser:myuser /

Re: [asterisk-users] Update peer IP address

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > > I do not want set allowguest=yes. The problem is, there is no official > > list with ip addresses of Telekom Germany. But I think all ip > >

Re: [asterisk-users] Update peer IP address

Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl : >Sebastian, > >If I have understood you correctly, the SIP communication is now via >NAT instead forwarded ports. For safety, it is much better. > >I think it is not because of a UDP timeout, but rather because of a

[asterisk-users] Re-Invite to Native Bridge

I installed Asterisk 13.6.0 hoping that I could get it to do this. But until now I haven't found out how. Does anybody know if this feature from the Media Format Rewrite article is already available? Kind regards, Sebastian -- ___

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

eck the SIP traces. You can either enable SIP debugging in Asterisk like so: sip set debug on Or you could run tcpdump and capture the SIP traffic. The first option is probably the easiest. Regards, Sebastian -- _ -- Bandwi

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

On Tue, Dec 22, 2015 at 09:30:52AM +, Luca Bertoncello wrote: > Zitat von Sebastian Kemper : > > Hi Sebastian > > > I tried with > > sip set debug 42 > sip set verbose 42 > > The result was in my first E-Mail... Hi Luca, I don't remem

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

to clean it up first (remove passwords, user names, phone numbers, digest authentication info etc). Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a li

[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?

o send out a contact URI in the register, that starts with sip: as well. Then inbound calls would work. Is there any way to get rid of this sips URI? Interestingly, when sending out calls, the Contact URI starts with sip instead of sips, so outbound calls

[asterisk-users] Asterisk registers with TLS, but sends out calls via UDP

k use the open TLS connection used for registering for outbound calls, too? Best Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory we

[asterisk-users] Queue grouping - how can it be implemented?

I have a Asterisk set up. In this, I want to use queues. Now I want to group "agents" into groups, such as so if one phone in a group is busy, the whole group is considered busy. Eg: Group1: SIP/Dad SIP/DadsMobile Group2: SIP/Mom SIP/MomsMobile If there is three persons in q

[asterisk-users] How can I "lock" a device or extension state to only specific states?

quot; device is that because its offline or not registred, then the person owning it can obviously not be engaged in the call, and thus its wise to ring the other, online device. Best regards, Sebastian Nielsen smime.p7s Description: S/MIME Cry

Re: [asterisk-users] Multiple readfile oddities, newlines etc

Theres always garbage in the end of the files. I do this when I want to read a file: same => n,Set(featurefile=/home/test/feature-1.txt) same => n,Set(unfilteredfeat2=${FILE(${featurefile},0,1,l,u)}) same => n,Set(feature2=${SHIFT(unfilteredfeat2)}) After that, add a , inside

[asterisk-users] Problems with REGEXP - anchor string to beginning

In extensions, I have this. The variable "oex" contains the original extension called, and is used to route outgoing calls internal or external depending on several factors. But now, im implementing a system that should require a passcode upon calling a "sensitive number". Here is the rele

Re: [asterisk-users] RS485 Audio device

Why RS485? Whats wrong with a simple 3-wire connection (monospeaker, monomic, ground) where you short monomic to ground on button press? Then you could use a simple usb device + device server to convert fron "smartphone headset" to usb then to network. On the server, you use a SIP phone client,

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

Im using SMS successfully over VoIP. No problems at all. You however need to use a good codec. However, I don’t use the MessageSend application, instead I use the raw SMS() application. This works by the SMS centre calling my fixed landline from a specific number, I detect the callerid, ini

Re: [asterisk-users] semi-OFF-TOPIC - SIP iptables and NAT - same source, different destination

Yes its called the state table. This because connection IP:PORT has a relationship with inside IP 192.168.x.x port X. I guess you have configured the redirect port to be same on both? Eg 5070 goes to *1:5060 and 5080 goes to *2:5060 What you need to do, is to have different inside ports a

Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

The 13.14 tar gz doesn’t even exists on the current or in the old releases folder. there seems to be an issue with the latest build not generating the artifacts? best regards On Feb 14, 2017, 11:04 -0300, Marcelo Terres , wrote: > Thanks Joshua. > Marcelo H. Terres IM: mhter...@jabber.mundoo

Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

is back online now thanks! On Feb 14, 2017, 11:18 -0300, Joshua Colp , wrote: > On Tue, Feb 14, 2017, at 10:13 AM, Sebastian Gutierrez wrote: > > The 13.14 tar gz doesn’t even exists on the current or in the old > > releases folder. > > > > there seems to be an issu

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

Check this one: https://github.com/IntegraCCS/integradesigner You can do many things, document each node, and save xml with each extension. We´ve made it open source on Astricon 2015 you can extend it the way you want. Hope it helps you. Best regards On Mar 18, 2017, 12:50 +0100, Jonathan H

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

on of .net it should be compiled with. > > The readme just points to the website. > > Thanks! > > On 18 March 2017 at 18:57, Sebastian Gutierrez wrote: > > Check this one: > > > > https://github.com/IntegraCCS/integradesigner > > > > You can do many thi

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

eed to know a few things like what > version of .net it should be compiled with. > > The readme just points to the website. > > Thanks! > > On 18 March 2017 at 18:57, Sebastian Gutierrez wrote: > > Check this one: > > > > https://github.com/IntegraCCS/integradesig

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

Everything is now on release folder on GitHub, documentation and executable. Hope it helps On Mar 18, 2017, 20:17 +0100, Sebastian Gutierrez , wrote: > This should work with at least .net framework 4, no dependency needed, just > .net framework, I think you should be able to compile it

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then yo

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

Use a callback. So when clocking in/out, they will hear a random 4 digit PIN, like "Enter four, three, six, eight at the callback". After they hangup, the phone will ring, and then they will have confirm with the 4 digit PIN. If they arent in presence: the phone at the site will ring, and the pers

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

l Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Nielsen Sent: Wednesday, May 10, 2017 2:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to detect f

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

Personally, if I was a client, I would rather have the personell answer the phone than make a outgoing call, if I would choose. If you think of billing and costs. So if a client allows outgoing, I don't think they have any problems with answering a call immediately following either. But I assume t

Re: [asterisk-users] Callee id over chan_sip trunk

I found very useful info here: https://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE In other words, on the asterisk1 box, you need to fetch from SIPPEER in extensions on asterisk1 box, and then populate connectedline. SIPPEER is the callee leg of the call, and CONNECTEDLINE is the calle

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

same here. > On Jun 12, 2017, at 10:02, Kseniya Blashchuk wrote: > > Same about me - need to re-enable membership all the time. Annoying (( > > пн, 12 июн. 2017 г. в 15:59, John Novack >: > Not just gmail > Happening as well with Comcast.net > > My Comcast address

Re: [asterisk-users] Writing CDR's to two database servers

use replication best regards > On Jun 19, 2017, at 17:47, Tech Support wrote: > > All; > I know that there are probably several solutions to this problem, but > what I am trying to do is provide some redundancy for my customers CDR data. > I know that doing simple backups of MySQL is prob

[Asterisk-Users] Media Negotiation Failed

Title: Mensaje Hi, I have this scenario   Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * )   When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and

[Asterisk-Users] RE: Media Negotiation Failed

L PROTECTED]> To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Media Negotiation Failed Date: Wed, 12 Nov 2003 13:01:29 -0300 Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0061_01C3A91D.16D9E3F0 Content-Type: text/plain; charset="iso-8859-1&q

[Asterisk-Users] Radius on *

Does Asterisk support Radius accounting? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-

[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs

ROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 2 Date: Mon, 17 Nov 2003 16:33:10 -0500 From: Jeremy McNamara <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Radius on * Reply-To: [EMAIL PROTECTED] Sebastian Nocetti wrote: >

[Asterisk-Users] G.723.1

Title: Mensaje Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that.   I have a GW 5300 and an ATA 186 and I want to place calls to PSTN.   I setup this config:   [general]port = 5060 bindaddr = xx.xx.xx.xx  context =

[Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS

hello all, I am having a trouble with Audio using h.323 channel...   I am doing this   Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING,

RE: [Asterisk-Users] chan_oh323

ldconfig, check that /etc/ld.so.conf have path to where oh323 library is   and then run ldconfig   De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Fathallah SoumayaEnviado el: Martes, 13 de Julio de 2004 12:27 p.m.Para: [EMAIL PROTECTED]Asunto: Re: [Asterisk-Users] chan_o

RE: [Asterisk-Users] How to uninstall Asterisk?

IN MY HONEST OPINION... IMHO I am right? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de ruixun wu Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] How to uninstall Asterisk? hi Gus and Roger,

[Asterisk-Users] CHAN_H323 bridge SIP no audio

I tried a lot of times to get it worked, but I cant obtain audio using SIP<->chan_h323 or chan_h323<->SIP   I tried disbling FastStart without good results...   What's the problem?   I need to do BRIDGE between SIP and H.323!!   help!!   Sebastian.-

RE: [Asterisk-Users] How to uninstall Asterisk?

it's also In My Humble Opinion too. Gonzalo P/D: Como andas Seba... :) On Wed, 2004-07-14 at 11:45 -0300, Sebastian Nocetti wrote: > IN MY HONEST OPINION... IMHO > > I am right? > > > -Mensaje original- > De: [EMAIL PROTECTED] > [mailto:[EMAIL PROT

[Asterisk-Users] Astersik with g729 and 120 active channels with digium card ISDN PRI

Hello, I want to know what kind of equipment I need to handle 120 simultaneous calls with a Digium 4E1 card... and using 120 G.729 licences some help?   thanks   Sebastian.

[Asterisk-Users] STILL NO AUDIO

I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING   It happened in both: SIP -> CHAN_H323 and CHAN_H323 -> SIP...   when it will be solved?

RE: [Asterisk-Users] STILL NO AUDIO

:39 a.m.Para: [EMAIL PROTECTED]Asunto: RE: [Asterisk-Users] STILL NO AUDIO Happen to have any NAT in the mix?   bkw   -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian NocettiSent: Monday, July 19, 2004 9:25 AMTo: [EMAIL PROTECTED]Subject

RE: [Asterisk-Users] STILL NO AUDIO

put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: > I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when > connected, NOTHING > > > > It happened in both: SI

RE: [Asterisk-Users] STILL NO AUDIO

Testing both... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Michael Manousos Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO Why don't you use asterisk-oh323? Michael. Seba

RE: [Asterisk-Users] STILL NO AUDIO

What kind of problem? All works OK except that config -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Holger Schurig Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO > I WANT TO USE

RE: [Asterisk-Users] Codecs - Advantages

If you dont have bandwith issues, use g711, with 2 mb bandwith you can pass 30 calls, aprox. G729 compress from g711 64 kbps to g729 8 kbps -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 19 de Julio de 2004 02:44 p.m.

RE: [Asterisk-Users] PSTN gateway implementation?

you can do that, in my experiencie, using oh323 I could not handle more than 30 active calls, doing g729 passthru...   I dont know how to do IP limitation for restrict ip access use iptables   I did basic dialpeers like this:   exten -> 1305.,1,dial(OH323/) exten -> 1305.,2,congest

RE: [Asterisk-Users] codec translate

To translate with g729 you need licenses... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Brent Franks Enviado el: Martes, 20 de Julio de 2004 10:01 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] codec translate > HI ALL; > > > Is astersik e

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