Re: [asterisk-users] SIP Source Port

Maybe it could be accomplished in the firewall? Tell the firewall to NAT the source port of packets to 5061? Från: asterisk-users-boun...@lists.digium.com För Alexander Perkins Skickat: den 10 juli 2021 19:39 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] SIP Source Port Hi

[asterisk-users] DECT client adapter

red into 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

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

Re: [asterisk-users] STIR/SHAKEN

an 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 reallt don’t under

Re: [asterisk-users] STIR/SHAKEN

as 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] 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"

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

om 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 accou

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

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 -- __

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

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 community forum at: https

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

Re: [asterisk-users] Channels freeze on Confbridge

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-users] Chann

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

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

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 nati

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

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

t 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 regards, Sebasti

[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

Re: [asterisk-users] Paging systems?

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 used wit

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:

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

. BTW: 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 -- Che

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

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 here

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, I don't

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

[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] 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

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

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

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

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

-Original 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 fak

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

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

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 <scg...@gmail.com>, wrote: > This should work with at least .net framework 4, no dependency needed, just > .net framework, I think you sh

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

too hard, I might need 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 <scg...@gmail.com> wrote: > > Check this one: > > >

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

at > 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 <scg...@gmail.com> wrote: > > Check this one: > > > > https://github.com/IntegraCCS/integradesigner

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

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 <jc...@digium.com>, 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. > > >

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

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

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,

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

[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

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

ailable" 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] Ast 13.10 to 13.11 stop working webrtc

the issue is fixed in current trunk head version El jue., 6 de oct. de 2016 a la(s) 12:07, Sebastian <scg...@gmail.com> escribió: > the issue is with chan_sip not on rtp I will check wich commit break this > and fill an issue. > > > El mié., 5 de oct. de 2016 a la(s)

Re: [asterisk-users] Ast 13.10 to 13.11 stop working webrtc

the issue is with chan_sip not on rtp I will check wich commit break this and fill an issue. El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian <scg...@gmail.com> escribió: > From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop > working, failing with > &g

[asterisk-users] Ast 13.10 to 13.11 stop working webrtc

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

[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

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

nnection 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 webinar every Thurs:

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

? Interestingly, when sending out calls, the Contact URI starts with sip instead of sips, so outbound calls work. Best Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?

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 <sebastian...@gmx.net>: > > Hi Sebastian > > > I tried with > > sip set debug 42 > sip set verbose 42 > > The result was in my first E-Mail... Hi

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

lean 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 live in

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

rian suggests to check 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 --

[asterisk-users] Re-Invite to Native Bridge

w I haven't found out how. Does anybody know if this feature from the Media Format Rewrite article is already available? Kind regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to A

Re: [asterisk-users] Update peer IP address

Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl <daniel.he...@gmail.com>: >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 tim

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] Asterisk 11 and pulseaudio setup as local user

Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis ge...@pagestation.com: 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

Re: [asterisk-users] Logging in local time

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://wiki.openwrt.org/doc/howto/voip.asterisk. Regards, Sebastian

Re: [asterisk-users] Signaling incoming call

basically find in this message: http://lists.digium.com/pipermail/asterisk-users/2015-April/286353.html Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Debugging dialplan

know the SMTP-daemon Exim... Is there such an option in Asterisk? Hi Luca, try 'dialplan show number@context'. Regards, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Debugging dialplan

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 lucab...@lucabert.de: Zitat von jg webaccounts...@jgoettgens.de: Yes, it is called core set

Re: [asterisk-users] Update peer IP address

). 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

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 had no changes any more. But we

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, found hundreds of servers

Re: [asterisk-users] Update peer IP address

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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Update peer IP address

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 is local behind a NAT with a firewall, following settings are used: externhost with DynDNS stun with stun.t-online.de http://stun.t-online.de

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

load = pbx_config.so load = app_cdr.so load = cdr_csv.so load = func_strings load = func_groupcount.so Any tips/hints/suggestions appreciated. Thanks for reading! Kind regards, Sebastian -- _ -- Bandwidth and Colocation

[asterisk-users] [PoE] Avaya 1152a1x

and no 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

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

in load, then 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

[asterisk-users] constantly increasing load in Asterisk 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 webinar

Re: [asterisk-users] Multicast AMI?

have you seen astmanproxy? best regards On Sep 23, 2014, at 10:05, jg webaccounts...@jgoettgens.de wrote: Hi! Maybe I have overlooked something, but I am sort of facing the following problem. I always used the AMI interface to allow (older) client programs on Windows to use their TAPI

Re: [asterisk-users] Asterisk and hylafax: how to debug ...

Am 08.05.2013 01:12, schrieb James Cloos: SN == Sebastian Niehaus nieh...@web.de writes: SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a SN virtual T.38 modem) for sending faxes. t38modem schould

[asterisk-users] Asterisk and hylafax: how to debug ...

]--- [general] static=yes writeprotect=no [1und1-fax-out] exten = _0.,1,Dial,SIP/${EXTEN}@495361000|45|r [default] include = 1und1-fax-out -[ end of extensions.conf ] Any idea what might be wrong? Thank you very much! Sebastian

Re: [asterisk-users] Asterisk and hylafax: how to debug ...

Am 07.05.2013 18:23, schrieb Sebastian Niehaus: For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. Well, I may add the log of t38modem (sorry for the ugly formatting) Parts I consider as most important are: ModemConnection::SetUpConnection dstNum=189659 srcNum

Re: [asterisk-users] Access postgresql directly from dialplan?

On 16/04/13 14:17, Gertjan Baarda wrote: On 16 apr. 2013, at 15:08, Sebastian Arcus s...@open-t.co.uk wrote: I would like to access a Postgresql database directly from my dialplan (to lookup names based on callerid numbers for incoming calls). Based on everywhere I looked - it seems the only

[asterisk-users] Access postgresql directly from dialplan?

I would like to access a Postgresql database directly from my dialplan (to lookup names based on callerid numbers for incoming calls). Based on everywhere I looked - it seems the only way to do this is with func_odbc. Considering that Asterisk seems to be able to access Postgresql databases

Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

at the bottom of this page a while ago regarding this issue: http://www.voip-info.org/wiki/view/chan_mobile At the time I also got hold of as many phones as possible and tested compatibility. I've added my results to the same page above. Sebastian

Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

On 31/01/13 10:15, Olivier wrote: 2013/1/31 Sebastian Arcus s...@open-t.co.uk mailto:s...@open-t.co.uk On 31/01/13 07:25, Olivier wrote: Hello, On a LAN, is it possible to install a bluetooth dongle on one workstation (at this time, this workstation OS

Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

in practice. It's quite likely that it will create too many problems, which will probably outweigh the benefits of what you are trying to do. Bluetooth already introduces a certain delay/latency in the communication path - by adding and IP link in between, that will only get worse. Sebastian

Re: [asterisk-users] Is there a need to secure RTP ports?

On 25/01/13 12:31, Johan Wilfer wrote: 2013-01-23 18:20, Sebastian Arcus skrev: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has

[asterisk-users] Is there a need to secure RTP ports?

I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). My question is - what are the vulnerabilities in

Re: [asterisk-users] Is there a need to secure RTP ports?

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus Sent: Wednesday, January 23, 2013 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is there a need to secure RTP ports? I have an Asterisk

Re: [asterisk-users] Is there a need to secure RTP ports?

On 23/01/13 17:33, Carlos Alvarez wrote: On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open

[asterisk-users] B200p card - use dahdi or mISDN?

I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when choosing which software to use? Is one better than the other - or does one have features

Re: [asterisk-users] B200p card - use dahdi or mISDN?

On 16/10/12 11:30, Patrick Lists wrote: On 10/16/2012 08:50 AM, Sebastian Arcus wrote: I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when

Re: [asterisk-users] B200p card - use dahdi or mISDN?

] On Behalf Of Patrick Lists Sent: 16 October 2012 12:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] B200p card - use dahdi or mISDN? On 10/16/2012 08:50 AM, Sebastian Arcus wrote: I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one

[asterisk-users] Question about async channel or macro for monitoring a call

Hi, Im trying to do this: 1) Originate a call between an external number and a ivr that do some things in background 2) after the originate I bridge the person that dial that extent with the external number I would like to have the ivr in background while the bridge is up for monitoring

Re: [asterisk-users] chan_mobile

://www.voip-info.org/wiki/view/chan_mobile Sebastian -- _ -- 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

Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

On 13/09/12 00:47, Vladimir Mikhelson wrote: On 9/12/2012 5:33 PM, Sebastian Arcus wrote: On 10/08/12 18:38, Chad Wallace wrote: On Tue, 31 Jul 2012 09:44:26 +0100 Sebastian Arcuss...@open-t.co.uk wrote: I have two setups with SIP hardware phones as extensions and POTS lines as trunks

Re: [asterisk-users] Trouble phoning via HUAWEI E169

},20,r) Also, what do you get when you run in Asterisk CLI: dongle show devices That should give you idea if the dongle is setup correctly in dongle.conf. Hope the above helps, Sebastian -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

On 10/08/12 18:38, Chad Wallace wrote: On Tue, 31 Jul 2012 09:44:26 +0100 Sebastian Arcuss...@open-t.co.uk wrote: I have two setups with SIP hardware phones as extensions and POTS lines as trunks. Internal SIP to SIP calls are crystal clear, but all calls bridged to POTS have a significant

[asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

is too high - by comparison with the quality on mobile phone calls (which are digital, incidentally) - so if I don't find a solution, I suppose I will just have to rip it all out and let one of the companies with proprietary phone systems install one. Any hints appreciated. Sebastian

[asterisk-users] audiohook errors

Hi, I´m facing some issues on asterisk 1.8.10. I can see this on the console: [May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: Operation not permitted [May 28 15:46:19] ERROR[28099]: lock.c:280

Re: [asterisk-users] Fax Problem on direct FXO port

get 70% of faxes ok. On May 18, 2012, at 1:35 AM, Steve Underwood wrote: Hi Sebastian, has still some issues that not all faxes pass ok, but does the work == still badly broken Your log doesn't seem to show a spandsp error. It looks more like a bad signal. Did you change anything

[asterisk-users] Fax Problem on direct FXO port

Hi, I´m with asterisk 1.6.2.20 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2 SpanDSP: spandsp-0.0.6pre20.tgz FXO lines. Sending faxes works ok. but receiving gives me error: here is the debug: http://pastebin.com/qfFeXWQW any idea?? Thanks! --

[asterisk-users] 1.8 busypatterns

Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length patterns, but

Re: [asterisk-users] 1.8 busypatterns

Can you point me to the commit to see if i can backport it? Thanks El 07/05/2012 18:50, Jonathan Rose jr...@digium.com escribió: - Original Message - From: Sebastian Gutierrez scg...@gmail.com To: asterisk-users@lists.digium.com Sent: Monday, May 7, 2012 10:38:03 AM Subject

Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

On 01/03/12 10:05, Sebastian Arcus wrote: I have a server with an OpenVox A400P card with 2 FXO modules on it. The internal extensions are SIP Grandstream phones. When making or receiving external calls through PSTN, there is an interrupted hissing like high pitch noise - which might go away

[asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

works fine. As I'm in the UK - I thought Europe uses alaw on phone lines - how come Asterisk seems to be using only ulaw - and alaw won't even work? Config examples on the Internet seem to suggest that it should work. Thanks for any help, Sebastian

Re: [asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

On 11/03/12 14:07, Tzafrir Cohen wrote: On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote: Hi all, I've tried to explicitly set my two PSTN trunks/FXO lines to alaw with: alaw = 1-2 in /etc/dahdi/system.conf. However, when I do this, all I get is loud intense noise on the line

Re: [asterisk-users] alaw = 1-2 in system.conf (dahdi) not working in UK

On 11/03/12 18:57, Shaun Ruffell wrote: On Sun, Mar 11, 2012 at 06:49:01PM +, Sebastian Arcus wrote: On 11/03/12 14:07, Tzafrir Cohen wrote: On Sun, Mar 11, 2012 at 12:05:48PM +, Sebastian Arcus wrote: Hi all, I've tried to explicitly set my two PSTN trunks/FXO lines to alaw

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