[asterisk-users] MACRO-INCOMING-CALL-TO-EXTENSION

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make

[asterisk-users] Crash in Asterisk

My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in

[asterisk-users] Asterisk going down

Hello my friends, My asterisk is going down randomly, following you will find some errors that i could see in the /var/log/asterisk/message at the moment of the crash: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202...@10.4.1.152 for seqno 21

Re: [asterisk-users] Losing local SIP phones when internet goes down?

Hello my friends, I'm having a problem like this post...the difference is that my asterisk goes down and i have to reboot my server in order to make it up again... following you will see some errors that i can see in the Asterisk /var/log/messages qhen asterisk goes down: [Feb 5 10:32:45]

Re: [asterisk-users] Asterisk going down

Thanks for the replay... I will check the memory tommorow with memtest86 from this site: http://www.linux.org/apps/AppId_7360.html I will let you know...but again, for me, the problem is in the network, may be some problem with the DHCP or DNS server from the client...what do you think? --

Re: [asterisk-users] Asterisk going dow

Thanks Josiah Bryan, I do not have any dns server running on my asterisk server, we do have an external DNS server working in the data center; the IP of this dns server is 10.4.1.5... Following you will see my main configuration: /etc/resolv.conf: domain localdomain search localdomain

Re: [asterisk-users] Asterisk going down (Josiah Bryan)

Thanks Josiah Bryan, I do not have any dns server running on my asterisk server, we do have an external DNS server working in the data center; the IP of this dns server is 10.4.1.5... Following you will see my main configuration: /etc/resolv.conf: domain localdomain search localdomain

Re: [asterisk-users] asterisk-users Digest, Vol 67, Issue 20 Re: Asterisk going down

requests? -- Josiah Bryan Productive Concepts, Inc. jbr...@productiveconcepts.com Cell: 765-215-0511 Desk: 765-215-6009 x224 -Original Message- From: Danny Dias ing.diasda...@gmail.com Date: Sun, 7 Feb 2010 15:48:19 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users

[asterisk-users] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds

Hello My friends, Today my asterisk stop working and i could see the following messags in /var/log/asterisk/messages at the time that asterisk stop working: [Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms / 2000ms) [Feb 16 13:24:41] NOTICE[8230] chan_sip.c:

[asterisk-users] Asterisk Crashs due to some Sip messages

Hello Asterisk community, Today my asterisk server stop working and i had to reboot the server in order to make it work again, take a look at the error messages in the CLI at the time of the crash: [Feb 25 12:44:20] WARNING[6965] chan_sip.c: sip_xmit of 0x920ae80 (len 545) to 10.4.2.3:5060

[asterisk-users] CLI not working properly - Asterisk Freez

Hello, I am using Asterisk 1.4.21.2 in a Centos 4.8 with a kernel version 2.6.9-89.ELsmp. The processor type is Intel(R) Xeon(R) Quad Core CPU E5410 @ 2.50GHz. with 4 GB of RAM Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls.

[asterisk-users] Asterisk DIES with no trace. PLEASE HELP!

Hello my friends We are having seriously problems with our asterisk server, our versions are as follows: WANPIPE Release: 3.4.7 Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.10.2 The symptoms are very weird, the CLI stop working suddenly, a core show channels shows MANY channels

Re: [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!

Thanks for the answer... No i did not make any changes to zapata, by the way, our problems began in December when we made a recompilation of everything in order to make an upgrade of libpri and wanpipe...just my guess :( maybe this is not the problem but just to let you know My /etc/resolv.conf

Re: [asterisk-users] Asterisk DIES with no trace. PLEASE HELP!

channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17 5:15 PM, Danny Dias ing.diasda...@gmail.com wrote: Thanks for the answer... No i did not make any changes to zapata, by the way, our

Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance Message: 10 Date: Thu, 18 Mar 2010 00:21:06 -0400 From: Zeeshan Zakaria zisha...@gmail.com

Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, SAngoma told me

[asterisk-users] How to make upgrades with Asterisk

Hello my friends, I want to make upgrades for all my software, currently i have the following versions: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 WANPIPE Release: 3.4.7 libpri version: 1.4.5 I want to make upgrade for the last version of Asterisk 1.4, the last version of Zaptel (dahdi will be

Re: [asterisk-users] How to make upgrades with Asterisk

Thanks Zeeshan, In fact,i have RealTime configured and working... What i want is to make an upgrade of libpri and wanpipe at least, asterisk and zaptel will be like i have now... Do you think that recompile/upgrade this softwares version will produce a problem? what steps should i do? Is it

[asterisk-users] Safe_asterisk doesn't exists???

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [r...@mypbx ~]# ps -A | grep

Re: [asterisk-users] Safe_asterisk doesn't exists???

: 626964fc1003240209i6a48bd5cj93fbacd25e2cf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 @Danny: How do you start your Asterisk ? -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 23 Mar 2010, Danny

[asterisk-users] Aastra weirds IP 169.x.x.x

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one:

Re: [asterisk-users] Safe_asterisk doesn't exists???

://en.wikipedia.org/wiki/Tux On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk

Re: [asterisk-users] Safe_asterisk doesn't exists???

://en.wikipedia.org/wiki/Tux On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know

[asterisk-users] Problems with Fax over TDM410P

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can

[asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes

Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we

Re: [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode

Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: x2saa4c40ff1004091730p192f37det33a5283a4ca85...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my

Re: [asterisk-users] Problems with Fax over TDM410P

! Message: 9 Date: Fri, 9 Apr 2010 19:22:05 -0430 From: Danny Dias ing.diasda...@gmail.com Subject: [asterisk-users] Problems with Fax over TDM410P To: asterisk-users@lists.digium.com Message-ID: y2l5a64fbaa1004091652k8393c88anf30c96809f8a9...@mail.gmail.com Content-Type: text/plain

Re: [asterisk-users] Asterisk DIES with no trace. PLEASE

Message-ID: 5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny

Re: [asterisk-users] Problems with Fax over TDM410P

, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com wrote: This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case

[asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)

? Please your help, we really need to put this working Thanks in advance for all your help! -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] incoming ghost call

this work ok, what should we do? Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] IBM X3650 with Asterisk???

to make this work better? Regards -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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 on Ubuntu

manually every time i reboot the machine (my laptop for testing) So, what should i do in order to solve this situation? Thanks in advance Regards -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- Bandwidth and Colocation

[asterisk-users] BLF with Realtime

Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1

[asterisk-users] WARNING[15867]: chan_sip.c:15766

Asterisk Dial Plan Hints =- 8...@pbx9: SIP/8340 State:Idle Watchers 0 - 1 hints registered And phone does not show any light with the the extension 8349 in use... Thanks in advance for your help -- Saludos Danny Dias

Re: [asterisk-users] BLF with Realtime

Thanks as always Zeeshan ;) I've changed my configuration, take a look: [8250] type=friend callerid=Extensión 8250 8250 canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250

[asterisk-users] BLF - Realtime Asterisk

' -= Registered Asterisk Dial Plan Hints =- 8340 at pbx9: SIP/8340 State:Idle Watchers 0 - 1 hints registered And phone does not show any light with the the extension 8349 in use... Thanks in advance for your help -- Saludos Danny Dias

Re: [asterisk-users] Maximum Wait Time queue option

Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos Danny Dias -- _ -- Bandwidth

[asterisk-users] How to finish an AGI

Hello community, I need to finish an AGI script when it invokes a macro from dialplan, how can i do that? it's quite confusing...the macro is making a hangup but the script continues Thanks -- Salu2 -- _ -- Bandwidth and

Re: [asterisk-users] How to finish an AGI

) exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden) What should i do to finish the macro if this macro reachs the Hangup? Thanks for your help my friend! 2010/9/2 Steve Edwards asterisk@sedwards.com On Thu, 2 Sep 2010, Danny Dias wrote: I need to finish an AGI script when

Re: [asterisk-users] How to finish an AGI

What should i do to finish the macro if this macro reachs the Hangup? I tried to say: What should i do to finish the *AGI* if this macro reachs the Hangup? 2010/9/2 Danny Dias ing.diasda...@gmail.com Hello Steven... Sorry for my poor explanation...what i'm trying to do is to invoke a Macro

Re: [asterisk-users] How to finish an AGI

Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias *Subject:* Re: [asterisk-users] How to finish an AGI snip This isn’t really a task for AGI since it is by nature single-call specific

Re: [asterisk-users] How to finish an AGI

YES YES...that's what i want ;) so simple but i was so tired :( I will try it and let you know ;) THANKS my friend 2010/9/2 Danny Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias

Re: [asterisk-users] How to finish an AGI

Any particular reason you don't want to put the logic of the macro in your AGI? Yes...i've no idea how to do it...it's a PERL script, i'm already checking how to do this...but it will be a little complicated :( 2010/9/3 Steve Edwards asterisk@sedwards.com On Thu, 2 Sep 2010, Danny Dias

Re: [asterisk-users] [SOLVED ]How to finish an AGI

); } # # By the way, is it necessary to Hangup the Macro if the AGI is already doing this? BR ;) 2010/9/3 Steve Edwards asterisk@sedwards.com On Thu, 2 Sep 2010, Danny Dias wrote: Sorry for my poor explanation...what i'm trying to do is to invoke a Macro from my AGI

[asterisk-users] How to avoid interruptions with DIGIUM

Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here:

Re: [asterisk-users] How to avoid interruptions with DIGIUM

Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we are using DELL servers, i've also used Sangoma, and always the same problem Thanks! 2010/9/9 Kevin P. Fleming kpflem...@digium.com On

Re: [asterisk-users] How to avoid interruptions with DIGIUM

please? Thanks! 2010/9/10 Moises Silva moises.si...@gmail.com On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias ing.diasda...@gmail.comwrote: Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we

Re: [asterisk-users] How to avoid interruptions with DIGIUM

Thanks Miguel, Excellent TIP! :) I will try and let you know Best Regards! 2010/9/10 Miguel Molina mmol...@millenium.com.co El 10/09/10 03:14, Danny Dias escribió: There used to be a problem with some Dell servers though, but that was already fixed some weeks ago. HEllo Moises

[asterisk-users] Problems with audio

Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check

Re: [asterisk-users] Problems with audio

Yes my friend...CONFIRMED!!! G729 on both sides 2010/9/15 Ishfaq Malik i...@pack-net.co.uk Have you checked that the codec order on the phone matched the order set on the server? On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: Hello, I'm having some problems with a total SIP

Re: [asterisk-users] Problems with audio

Hello Adriá... We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost 1000 users, we've checked the gain and volume on the phones :( 2010/9/15 Adrià Vidal adriavi...@gmail.com On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.comwrote: Yes my friend

Re: [asterisk-users] Problems with audio

Thanks Sebastian, It's the same firmware version for all our linksys phones...and we have hundreds of pbx's runnning this firmwares versions without any problem 2010/9/15 Sebastian s...@open-t.co.uk Hi, On 09/15/2010 04:04 PM, Danny Dias wrote: Hello, I'm having some problems

[asterisk-users] How to Understand a pri intense debug span X

Hello my friends, I would like to understand the output from pri intense debug span X, the Telco always says that their side is OK, but i always receive alarms, loosing connection, take a look: [Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1: Recovering [Sep 16 13:18:19]

Re: [asterisk-users] How to Understand a pri intense debug span X

Any hints please? I would appreciate your valuabl help Thanks 2010/9/16 Danny Dias ing.diasda...@gmail.com Hello my friends, I would like to understand the output from pri intense debug span X, the Telco always says that their side is OK, but i always receive alarms, loosing connection

[asterisk-users] How to test BIG traffic through DAHDI/WANPIPE interfaces

Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD --

Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

*Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i

Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar

[asterisk-users] Problems compiling Asterisk on Debian

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed.

Re: [asterisk-users] Problems compiling Asterisk on Debian

installed on every install. This link should help: http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/ Dean Hoover Milwaukee, Wisconsin On 9/27/2010 11:09 AM, Danny Dias wrote: Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: r...@sangoma-testing

Re: [asterisk-users] Problems compiling Asterisk on Debian

installed. You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 The same result :( 2010/9/27 Daniel Tryba dan...@tryba.nl On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4

Re: [asterisk-users] Problems compiling Asterisk on Debian

¿ahhh? 2010/9/27 Roger Burton West ro...@firedrake.org On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: What should i do? aptitude install module-assistant m-a a-i dahdi -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Problems compiling Asterisk on Debian

I've these versions of DAHDI running into another Server with no problem...it seems to be a problem with dependencies, but i can't find the trick :( 2010/9/27 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com wrote: What should i

Re: [asterisk-users] Problems compiling Asterisk on Debian

paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com wrote: What should i do? Try with the lastest DAHDI version, 2.4.0. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode

Re: [asterisk-users] Problems compiling Asterisk on Debian

-la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux - linux-headers-2.6.26-2-amd64 Seems to be OK, isn't? Thanks! 2010/9/27 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com wrote: The same problem! What

Re: [asterisk-users] Problems compiling Asterisk on Debian

./configure was run? If so redo the ./configure command and see what that does. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home

Re: [asterisk-users] Problems compiling Asterisk on Debian

: cannot access /usr/src/linux: No such file or directory Is that Ok? 2010/9/28 Danny Dias ing.diasda...@gmail.com Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root

Re: [asterisk-users] Problems compiling Asterisk on Debian

-a Linux Sangoma-Testing 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 2010 x86_64 GNU/Linux r...@sangoma-testing:/home# uname -r 2.6.26-2-amd64 2010/9/28 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias ing.diasda...@gmail.com wrote: r...@sangoma-testing

Re: [asterisk-users] NAT issue (i think?)

You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W

[asterisk-users] What's the meaning of this?

Hello, I'm checking this: [Sep 28 13:32:46] NOTICE[30360] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event:

Re: [asterisk-users] NAT issue (i think?)

:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias

[asterisk-users] Weird Behavior with DAHDI

Hello, I'm experiencing some weird problems on my server: - 1) The following messages are filling up my logs: [Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Sep 29 08:24:59] WARNING[7078]:

[asterisk-users] Kernel Panic When restarting the server

Hello, I'm getting a KErnel Pannic every time i restart the server, what could be happening? I just make: shutdown -r now and the server gets Kernel Panic. I'have to go on site and press the power button Here you have my sotware versions: Asterisk 1.4.24.1 DAHDI Tools Version - 2.1.0.2 DAHDI

Re: [asterisk-users] Kernel Panic When restarting the server

Thanks Tim That solved my problem, thank you very much...but now i'm having another problem, when the server starts, it doesn't start asterisk automatically, should i change the start script? 2010/9/30 Tim Nelson tnel...@rockbochs.com - Danny Dias ing.diasda...@gmail.com wrote: I'm

Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello

Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.comwrote: Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall

[asterisk-users] checking CDR

Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never

Re: [asterisk-users] SIP 401

Zakir, Have you checked the RFC3261? 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 2010/10/20 Zakir Mahomedy z...@mayfair2000.com Hi I am trying to get

Re: [asterisk-users] SIP 401

[:port][/extension] 2010/10/20 Danny Dias ing.diasda...@gmail.com Zakir, Have you checked the RFC3261? 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 2010

[asterisk-users] dialing from asterisk console?

Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By

[asterisk-users] res_musiconhold.c Bug - Patch to solve?

not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Ing. Danny Dias www.DannTEL.net

Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?

this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Dog is my Co-pilot -- Ing. Danny Dias

Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?

2010/11/22 John Novack jnov...@stromberg-carlson.org Danny Dias wrote: Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? I can't answer that question. ummm why

[asterisk-users] Asterisk + Phones behind different Nat Firewalls

Hello, I have a doubt (basic i guess, but not for me). I have an escenario where customer site has Asterisk PBX behind Nat/firewall with private IP address and sone phones also; BUT there are some other phones on different sites and of course behind its nat/firewalls; with IAX i have no problem,

Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls

behind nat El 26/04/2012 19:31, Carlos Alvarez car...@televolve.com escribió: On Thu, Apr 26, 2012 at 9:54 AM, Danny Dias ing.diasda...@gmail.comwrote: I have a doubt (basic i guess, but not for me). I have an escenario where customer site has Asterisk PBX behind Nat/firewall with private IP

Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls

Does not work for me! El 26/04/2012 20:14, Carlos Alvarez car...@televolve.com escribió: On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias ing.diasda...@gmail.comwrote: I cant put public ip adress to the asterisk server. The main problem i see is with the sip headers (contact, sdp ip and ports

Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls

Ok understood. The signaling wont be a problem, but not the same with rtp as it uses randomly ports. The idea is to have an intermediary who could delivers both ports and ping them to both sides to keep nating open on routers, this is what i do with rtp proxy within opensips. But in this case no

Re: [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)

Did you asked OpenVOX for support? El 27/04/2012 01:48, John Millican j...@millican.us escribió: Hello, I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia, USA. Whenever I place a call to one of the two

Re: [asterisk-users] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)

Btw, red alarms means phisical problemscheck cable first. El 27/04/2012 10:23, Danny Dias ing.diasda...@gmail.com escribió: Did you asked OpenVOX for support? El 27/04/2012 01:48, John Millican j...@millican.us escribió: Hello, I have an OpenVox A400E02 (2FXO) in a box running Debian

Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls

on the firewall also you should set the phone to send a nat keep alive each 30 seconds (asterisk also sends a options packet to keep the nat open but doesn't always work ok ) -Original Message- From: Danny Dias ing.diasda...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date

[asterisk-users] Digium IP Phones

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull

[asterisk-users] R-Series with NON-DIGIUM card on servers

Hi, I would like to know if the servers (A and B) could use boards non-digium with the R-Series HA product from Digium, i have a couple of B600E Sangoma to put on each server and use the R-series to provide HA. Is that possible? Thanks -- www.danntel.net *sip:danny4...@thesipschool.com*

Re: [asterisk-users] Digium IP Phones

Does the D40 will support the option to develope apps? As i could see on videos only the D70 has the apps button, and also, the lcd screen is smaller. Right? Enviado desde mi Samsung Galaxy S II El 10/05/2012 12:44, Kevin P. Fleming kpflem...@digium.com escribió: On 05/09/2012 08:38 PM, Danny

Re: [asterisk-users] R-Series with NON-DIGIUM card on servers

Thanks, What about the Database and recording calls replication? as i could see, the RSeries does not take into account these data. Thanks 2012/5/12 Kevin P. Fleming kpflem...@digium.com On 05/11/2012 10:46 PM, Danny Dias wrote: Hi, I would like to know if the servers (A and B) could use

Re: [asterisk-users] R-Series with NON-DIGIUM card on servers

Thanks Kevin. Buying one for Spain right now ;) 2012/5/15 Kevin P. Fleming kpflem...@digium.com On 05/12/2012 12:07 PM, Danny Dias wrote: What about the Database and recording calls replication? as i could see, the RSeries does not take into account these data. The Digium R-series

[asterisk-users] DPMA for Digium Phones

Hello, I have a question regarding DPMA for Digium Phones, if i install the DPMA on my Asterisk Server A, and then, i move the phone to register into another Asterisk Server B, can i install for free another DPMA license for my digium phones on this second server? can i move the DPMA from one

Re: [asterisk-users] DPMA for Digium Phones

2012/5/21 Danny Dias ing.diasda...@gmail.com Hello, I have a question regarding DPMA for Digium Phones, if i install the DPMA on my Asterisk Server A, and then, i move the phone to register into another Asterisk Server B, can i install for free another DPMA license for my digium phones

[asterisk-users] Deleting OLD Voicemails

Hello, I was checking how to DELETE old voicemail from Asterisk, for my extension 300, i have 20 MB [root@pbx INBOX]# pwd /var/spool/asterisk/voicemail/default/300/INBOX [root@pbx INBOX]# du -s -h 20M There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV I've

Re: [asterisk-users] Deleting OLD Voicemails

Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav

Re: [asterisk-users] Deleting OLD Voicemails

23, 2012, at 1:03 AM, Danny Dias wrote: Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail

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