[Asterisk-Users] Bad soundquality on inbound calls.

it would look somethnk like this. IP Phone - Asterisk - PSTN Kind Regards, Jakob ### Detta mail har blivit skannat av F-Secure Anti-virus for Microsoft Exchange. For mer information, ga till http://www.F-Secure.se This message has been scanned by F

[Asterisk-Users] Problem with inbound call quality.

I wonder where I should start looking for problems when my symptoms are: * Good quality on outbound (X-lite - asterisk - PSTN via X100P) calls. * Bad quality, very low volume and some distortion, on outbound (PSTN - asterisk - X-lite via X100P) calls. Regards, Jakob

Re: [Asterisk-Users] problem to place calls to NIKOTEL

the wrong call setup packet. The not working Dial command did not take the variables form the context nikotel in the sip.conf, because this context was not called. I was a bit confused by the examples. jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] who has German voice files ?

Hi, I like that my * talks German to the callers. Google does not give me any reference about the availability of german announcement files. Could somebody on this list help me out and make it available to me. Thanks, best regards Jakob

Re: AW: [Asterisk-Users] who has German voice files ?

Thomas, At 14:45 11.03.2004 +0100, you wrote: Wait a week and you can have german files from one of our customers, who wants to donate such files. Please let us know when they are available. Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] Music on Hold sound goes off if environment is silent

Hi, Music on hold works if the environment is noisy. But in case of silence the sound goes off. If I scratch continuously on the mikrofone, then the replay works without any interruption. Q: is there a parameter which influences this behaviour? Thanks, best regards Jakob

Re: [Asterisk-Users] Music on Hold sound goes off if environment is silent

Hank, can you play music on hold using the line in feature of your sound card to the phone? I have a Logitech USB Headset, which has integrated Sound Card. I cant find the line feature, can you give me a hint where to find it? Jakob BTW: the silence suppression as a workaround is working

RE: [Asterisk-Users] How to send CallerID trough CAPI ?

Florian, Thanks for your help. I have tried your version but the result is still the same. It seems that I make a fundamental error. Can you bring me back on the way ? Jakob For CAPI you have to set the CallerID in your DialString: exten = _0800XXX.,1,SetCIDNum(${CLID}) exten = _0800XXX.,2

RE: [Asterisk-Users] How to send CallerID trough CAPI ?

Florian, Thanks, big step in the right direction. But the Called User sees now 062775170 wich is the root number (or the initial number where my ISDN numbering starts) The CLID is 0627775171 which is defined in my [globals] Jakob asterisk*CLI -- Executing SetCIDNum(SIP/1234-d016

RE: [Asterisk-Users] How to send CallerID trough CAPI ?

Florian, Thanks, this is a step back. I have tried now almost all possible dial combinations but no luck. The result of your suggestion is at the end of the mail. BTW: is it also raining in Holland? Jakob -Original Message- big step in the right direction. But the Called User sees

Re: [Asterisk-Users] RH 9 with AVM C2 ISDN - New User - Guru's Help needed Urgent!!

was able to help at least jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] How to send CallerID trough CAPI ?

Florian, Thanks. Now it works. jakob The correct dial syntax for CAPI channels is like this: CAPI/12345678:b${EXTEN} where: 12345678 is your outgoing MSN (you would choose 0627775171) and ${EXTEN} is the number to dial. My mistake was I moved the 'b' too when I switched the two numbers around

Re: [Asterisk-Users] Guru's help with * and AVM C2 ISDN - Newbie going mad!!

I would make the following change in extensions.conf ( I I had to do it this way in my extensions.conf, even the context name in capi.conf was different. I do not understand why but it works) hope this helps Jakob [default] include - demo [demo] ; ; We start with what to do when a call first

Re: [Asterisk-Users] ISDN BRI with DDI support

? As far as I remember I heard once that the active DIVA Cards do echo cancellation. Can anybody comment on this? Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

- make install - reboot my machine - modprobe capi - insmod -f fcpci What do I miss? What did I do wrong? regards jakob asterisk:~# lsmod Module Size Used by Tainted: PF fcpci 532320 2 capi 6528 4 kernelcapi 30624 3 [fcpci capi

Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

suggested in the HOWTO. jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

, but what where ? Thanks in advance for your help jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Asterisk and Sip/IP Phones

Mark, At 10:23 03.06.2004, [EMAIL PROTECTED] wrote: I mean the: GrandStream BT-101 This is what I am using. I have also a 7960. Voice quality wise the BT-101 does not stay behind the later Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] FWD network from Asterisk through NAT

to the Internet (leased line) or in my case I am using dynamic DNS with the ADSL Service. Below is the relevant part of SIP.CONF Jakob ; SIP Configuration for Asterisk ; ; SIP.CONF ; [general] nat=yes ; externip = abcde.dyns.net ; Addr put in SIP messages if we're behind a NAT

Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

Thomas, Thank you for your help. Which Kernel version do you run? I have debian 2.4.24, may be I could use your patched object code? Could you please send it to me if it makes sense. My primary mail is jakob-at-teamstrebel.ch Doing it this way would make my second ISDN line operational. Later I

Re: [Asterisk-Users] Help needed (New to Asterisk)

with kernel 2.4.24 and on Redhat 9.0. PC's are 1GHz 512MB RAM. The System is slightly loaded Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Re: MOH doesn't play

. Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Strange SIP issue (again)

Andreas, below is my partial sip.conf (which is relevant for fwd) this works for me. jakob [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to nat=yes ; externip = myhost.dyns.net ; Addr put in SIP messages if we're behind

Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

://www.junghanns.net/asterisk/page1.html I have this working. jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] Re: [Asterisk-Users] Réf. : Re: [ Asterisk-Users] Fritz ISDN PCI v2 and CAPI

, by the way...) I remember that I read (found trough Google) that hisax and channel_capi can not be installed at the same time. Actually you have to remove hisax to run chan_capi jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

RE: [Asterisk-Users] Permanently logged in agents?

PROTECTED] On Behalf Of Peter Svensson Sent: Friday, September 17, 2004 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Permanently logged in agents? On Fri, 17 Sep 2004, Jakob Borg wrote: I'm looking for a way to do queue management

[Asterisk-Users] Permanently logged in agents?

Hi, I'm looking for a way to do queue management for several phones. Basically I want the phones to act as normal, except that incoming calls get queued if the phone is busy at that moment. This can be achieved with one queue per extension that wants this behavior (and actually works well like

[asterisk-users] 1.8.2.4: SIP dialogs not killed?

Hi, I'm wondering if this is normal asterisk behaviour: asterisk*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing)No Rx: PUBLISH

Re: [asterisk-users] 1.8.2.4: SIP dialogs not killed?

, Johannes Jakob wrote: Hi, I'm wondering if this is normal asterisk behaviour: asterisk*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing

Re: [asterisk-users] 1.8.2.4: SIP dialogs not killed?

Hi, On 28.02.2011, at 22:55, Danny Nicholas wrote: If it is affecting your system performance, post again (and try to use your nice voice :) ). sorry, I really didn't want to be unfriendly. English isn't my mother tongue, so I might get the wrong tone sometimes... sorry for that. On

Re: [asterisk-users] Binding asterisk to two static IPs

On 12.10.2011 23:27, ge...@riseup.net wrote: If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 ... Any idea how to solve this? Yes: Use

Re: [asterisk-users] Binding asterisk to two static IPs

On 13.10.2011 00:27, ge...@riseup.net wrote: If I use the floating internal ip, I can't reach my provider anymore. Thought this was clear. After reading your original message, this is clear, yes. Sorry for being sloppy. --

Re: [asterisk-users] Binding to 0.0.0.0 a security risk?

Steve Edwards, 2012-02-06 01:43: Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. Right, that's what I also observed. We had to take special measures to handle this. The problem lies in the nature

Re: [asterisk-users] Binding to 0.0.0.0 a security risk?

Raj Mathur (राज माथुर), 2012-02-08 03:27: Packets not going out on the same interface as the one they were received on is a general IP issue, not just for connectionless Right, this was a inaccuracy. It should say Asterisk does not reply with the IP address with which packets were received.

[asterisk-users] IP address of remote SIP host

Hi, is it possible to get the SIP IP address of the remote (calling) party, in the dialplan or (preferrably) in an AGI script? (This sounded like a rather basic question to me, but I could not find an answer...) TIA regards Jakob

[asterisk-users] DTMF inband with telephone-event in SDP

behaviour is even RFC compliant? Regards and TIA, Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Asterisk 1.8 and dual stack support

Jaap Winius, 21.03.2013 17:47: support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only friends and peers will be able to connect

[asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

. Saving audio to a file is disabled. Copyright (C) 2000-20012, Cepstral LLC. Do You have any Ideas why that won't work? Best Regards Jakob Böttger smime.p7s Description: S/MIME Kryptografische Unterschrift

Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

Am 20.06.2012 14:24, schrieb Darren Sessions: Hi Jakob, I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). - D On Jun

[asterisk-users] Queue Member login from IAX trunk

-newlocation. Do you have any ideas how to solve that? Best regards Jakob smime.p7s Description: S/MIME Kryptografische Unterschrift -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Queue Member login from IAX trunk

-newlocation) exten = 105,n,Set(AGENT_SIP=${DB(IAX2/intranet/agent_ip)}) so the agent enters the number from the phone he is connected. Then Asterisk adds IAX2/serverb to the number and saves it as agend phone number... Regards Jakob Am 04.07.2012 11:45, schrieb SamyGo: Hi, exten = 105,n,Read

[asterisk-users] Asterisk Fax detection *11.7

it until answer for SIP/abcde-0016 -- SIP/123-0018 is ringing -- SIP/456-0017 is ringing Any hints why thats not working? Best Regards Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk Fax detection *11.7

-0017 is ringing is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why? Regards Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Asterisk Fax detection *11.7

it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de mailto:ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 tel:%5B12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack

Re: [asterisk-users] Asterisk Fax detection *11.7

Jakob -- _ -- 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 mailing list

[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

(error) : HANGUP) in new stack any hints? Best Regards Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Thanks Am 03.02.2014 11:57, schrieb Larry Moore: On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942

[asterisk-users] Asterisk 11.11 with TCP/TLS SRTP and Grandstream gxp1450 not working

Jakob signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success

i'm using asterisk with tls but always get WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success whats wrong there? Best Regards Jakob signature.asc Description: OpenPGP digital signature

[asterisk-users] SRTP only from asterisk to extention possible

=friend host=dynamic transport=tls,tcp qualify=yes directmedia=no [200](NAT) callerid=200 defaultuser=200 fromuser=200 secret=password mailbox=200@default Best Regards Jakob signature.asc Description: OpenPGP digital signature