[asterisk-users] Google Puts the Final Nail in the Google Voice Coffin
No tell me that's a jock ! I can't believe it: http://nerdvittles.com/?p=7940 -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No voice when the calls come from Internet
Hi, I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don't hear any thing! I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP range in rtp.conf forwarded to my Asterisk server. Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ? Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call. Regards, -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=1 rtpend=10100 And My FW is setup to forward all incoming ports of range 1-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens Subject: Re: [asterisk-users] RTP port ranges To: Andrew Colin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <5232dfc7.2030...@telenet.be> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: > Because normally it will use a random port between them > > On 9/13/2013 11:43 AM, Jonas Kellens wrote: >> On 09/13/2013 11:41 AM, Andrew Colin wrote: >>> Normally you should open ports 1-2 udp >>> >>> >>> >>> On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. >>> >> >> >> Why do I need such a big range ? That's like for 250 concurrent calls ! >> >> >> >> Jonas. >> > -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use Skype ?
Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? Thanks -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Joining an astablished call
Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu It's just a normal call between to channels that I have to join for few minutes. Regards -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to show caller number ?
Hi, I am using asterisk 11.1.0. How to display the caller number (from asterisk -rvvv terminal) in the first step of the extension (before doing any action) ? Thanks -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] weird "RED alarm" on FXO channel
I have a recurrent problem on my asterisk box. I have "VIA Samuel 2" as a CPU. With asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled from source. I get a RED alrm drom the port 1( FXO) two or three times per day: [Feb 4 15:54:57] WARNING[9991]: chan_dahdi.c:8018 handle_alarms: Detected alarm on channel 1: Red Alarm I plug the RJ11 in a normal phone and I can hear the line is up and I can call extern number (via PSTN) I must stop asterisk + restart dahdi and start asterisk to get asterisk work normaly. I have no idea where is the problem or even what to do to debug the issue I get these outputs when the channel is in "RED Alarm" root@pbx01:~# /etc/init.d/dahdi status ### Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER) 1 FXOFXSKS (In use) (EC: MG2 - INACTIVE) RED < This is my problem ! 2 FXOFXSKS (EC: MG2 - INACTIVE) RED 3 FXSFXOKS (In use) (EC: MG2 - INACTIVE) 4 FXSFXOKS (In use) (EC: MG2 - INACTIVE) pbx01*CLI> dahdi show status Description Alarms IRQbpviol CRCFra Codi Options LBO Wildcard TDM410P OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) pbx01*CLI> pbx01*CLI> dahdi show channels Chan Extension Context Language MOH InterpretBlocked State Description pseudodefaultdefault In Service 1from-pstn default In Service 3local default In Service 4local default In Service pbx01*CLI> dahdi show cadences channel channels statusversion pbx01*CLI> dahdi show cadences r1: 125,125,2000,4000 r2: 250,250,500,1000,250,250,500,4000 r3: 125,125,125,125,125,4000 r4: 1000,500,2500,5000 pbx01*CLI> dahdi show version DAHDI Version: 2.6.1 Echo Canceller: HWEC, MG2 pbx01*CLI> dahdi show channel 1 Channel: 1 Description: File Descriptor: 12 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID subaddress: Caller ID name: port-bell01 Mailbox: 1 Destroy: 0 InAlarm: 1 Signalling Type: FXS Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps currently OFF Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Is it a known bug ? Why I have to restart dhadi to resolve this ? Thanks a lot ! -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Build asterisk for VIA C3
Is it difficult to publish a build asterisk.deb compiled for VIA C3 architecture ? Instead of using the binary just for me. So any one trying to install it on C3 CPU will need just to do: aptitude install asterisk The one that is installed by default doesn't work for such a CPU Should I contact debian dev team for that? Thanks OLD messages --- Message: 6 Date: Mon, 31 Dec 2012 12:08:40 -0500 From: neo haux Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: asterisk-users@lists.digium.com Message-ID: Content-Type: text/plain; charset=ISO-8859-1 Thanks George it works now ! Message: 6 Date: Sun, 30 Dec 2012 17:18:45 -0700 From: George Joseph Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset="utf-8" Try this... In menuselect, uncheck BUILD_NATIVE under Compiler Flags and recompile. On Sun, Dec 30, 2012 at 4:44 PM, neo haux wrote: > Hi, > > I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2 > 800MHz CPU. A small box to play with PBX at home. > > I get this error when I start asterisk: > > root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start > Illegal instruction > Starting Asterisk PBX: asteriskIllegal instruction > > I compiled it on debian 6.0.6 with this options: > ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \ > --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \ > --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib > > > I was able last days to compile asterisk 1.8 and it did work > perfectly except with Gtalk, and this is why I have to compile > v11.1 -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
Thanks George it works now ! Message: 6 Date: Sun, 30 Dec 2012 17:18:45 -0700 From: George Joseph Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset="utf-8" Try this... In menuselect, uncheck BUILD_NATIVE under Compiler Flags and recompile. On Sun, Dec 30, 2012 at 4:44 PM, neo haux wrote: > Hi, > > I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2 > 800MHz CPU. A small box to play with PBX at home. > > I get this error when I start asterisk: > > root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start > Illegal instruction > Starting Asterisk PBX: asteriskIllegal instruction > > I compiled it on debian 6.0.6 with this options: > ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \ > --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \ > --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib > > > I was able last days to compile asterisk 1.8 and it did work > perfectly except with Gtalk, and this is why I have to compile > v11.1 -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
Hi, I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2 800MHz CPU. A small box to play with PBX at home. I get this error when I start asterisk: root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start Illegal instruction Starting Asterisk PBX: asteriskIllegal instruction I compiled it on debian 6.0.6 with this options: ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \ --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \ --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib I was able last days to compile asterisk 1.8 and it did work perfectly except with Gtalk, and this is why I have to compile v11.1 I have gone through these steps (v1.8): 1) ./configure --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib 2) in Makefile change : PROC=i586 (instead of i686) 3) in makeopts change: #BUILD_PLATFORM=i686-pc-linux-gnu BUILD_PLATFORM=i586-pc-linux-gnu #BUILD_CPU=i686 BUILD_CPU=i586 #HOST_PLATFORM=i686-pc-linux-gnu HOST_PLATFORM=i586-pc-linux-gnu #HOST_CPU=i686 HOST_CPU=i586 PROC=i586 4) # make && make install But I can't do the same changes for asterisk 11.1 because the makefiles haven't the same content as for asterisk v1.8 I need asterisk 11.1 to be able to receive and make calls to gtalk accounts Do you have any idea where is the problem ? My VIA CPU: root@pbx01:/usr/src/asterisk-11.1.0# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping: 3 cpu MHz : 799.892 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr cx8 mtrr pge mmx 3dnow bogomips: 1599.78 clflush size: 32 cache_alignment : 32 address sizes : 32 bits physical, 32 bits virtual power management: -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: asterisk-users Digest, Vol 95, Issue 33
Thanks I had this line in my /etc/asterisk/chan_dahdi.conf :
include=/etc/asterisk/dahdi-channels.conf
the file /etc/asterisk/dahdi-channels.conf was generated by
/usr/sbin/dahdi_genconf
I simply did that :
cat /etc/asterisk/dahdi-channels.conf >> /etc/asterisk/chan_dahdi.conf
It works now.
May be the option "include" is not supported within the file chan_dahdi.conf
--
Message: 2
Date: Sat, 23 Jun 2012 16:34:35 -1000
From: Julian Yap
Subject: Re: [asterisk-users] Can't make call with TDM410P
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID:
Content-Type: text/plain; charset=ISO-8859-1
On Sat, Jun 23, 2012 at 10:32 AM, neo haux wrote:
> Actually I can start and receive SIP calls (PC client, iphone client)
> but?I have an issue with calling external number throught PSTN
> (certified-asterisk-1.8.11-cert2).
>
> I'm having this ?error when making a call:
>
> *CLI> ? == Using SIP RTP CoS mark 5
> ? ? -- Executing [9000@local:1] Dial("SIP/3000-0006",
> "DAHDI/1/4384019357,10") in new stack
> [Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
> ? == Everyone is busy/congested at this time (1:0/0/1)
> ? ? -- Executing [9000@local:2] Hangup("SIP/3000-0006", "") in new stack
> ? == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-0006'
>
>
> My configs :
> *CLI> dahdi show channels
> ? ?Chan Extension ?Context ? ? ? ? Language ? MOH Interpret
> Blocked ? ?State
> ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default
> ?? ? ? In Service
Where are your channels? That's why you are receiving the error
"Unable to create channel of type 'DAHDI'".
Define your channel groups in /etc/asterisk/chan_dahdi.conf
Then is should look like this:
# asterisk -rx "dahdi show channels"
Chan Extension Context Language MOH Interpret
Blocked State
pseudo default default
In Service
1 from-external en default
In Service
2 from-external en default
In Service
3 from-external en default
In Service
4 from-external en default
In Service
5 from-external en default
In Service
6 from-external en default
In Service
...
- Julian
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[asterisk-users] Can't make call with TDM410P
Actually I can start and receive SIP calls (PC client, iphone client)
but I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2).
I'm having this error when making a call:
*CLI> == Using SIP RTP CoS mark 5
-- Executing [9000@local:1] Dial("SIP/3000-0006",
"DAHDI/1/4384019357,10") in new stack
[Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9000@local:2] Hangup("SIP/3000-0006", "") in new stack
== Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-0006'
My configs :
*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
Blocked State
pseudo default default
In Service
*CLI> dahdi show status
Description Alarms IRQ bpviol CRC
Fra Codi Options LBO
Wildcard TDM410P OK 0 0 0
CAS Unk 0 db (CSU)/0-133 feet (DSX-1)
*CLI>
root@my-PC:/usr/src/certified-asterisk-1.8.11-cert2# lsmod | grep dahdi
dahdi_echocan_mg2 12998 4
dahdi_voicebus 58608 1 wctdm24xxp
dahdi 220595 3 dahdi_echocan_mg2,wctdm24xxp,dahdi_voicebus
crc_ccitt 12667 2 wctdm24xxp,dah
extensions.conf
[local]
exten => 100,1,Dial(gtalk/asterisk/myaccount.v...@gmail.com)
exten => 2000,1,Dial(SIP/2000,10)
exten => 3000,1,Dial(SIP/3000,10)
exten => 9000,1,Dial(DAHDI/1/MyCellPhoneNumber,10)
exten => 9000,2,hangup()
root@My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_scan
[1]
active=yes
alarms=OK
description=Wildcard TDM410P
name=WCTDM/0
manufacturer=Digium
devicetype=Wildcard TDM410P
location=PCI Bus 04 Slot 01
basechan=1
totchans=4
irq=0
type=analog
port=1,FXO
port=2,FXO
port=3,FXS
port=4,FXS
root@My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_hardware
pci::04:00.0 wctdm24xxp+ d161:8005 Wildcard TDM410P
--
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Re: [asterisk-users] Delay before ringing from PSTN`s call
Hi,
I commented the option callerid in the file dahdi-channels.conf without
success, My SIP phone still ring after 4-5 secondes :-(
; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"
signalling=fxs_ks
;callerid=asreceived
I am living in Canada, so I guess I should use USA signaling ? If so in
which file ?
Message: 7
Date: Tue, 04 Oct 2011 14:49:55 -0400
From: John Novack
Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: neo haux
Message-ID: <4e8b5553.2030...@stromberg-carlson.org>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your
country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper
configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protocols
You will need to dig into that first.
John Novack
neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
>
> I configured incoming calls from pstn to ring my SIP phone (extension :
100)
>
> cat extensions.conf
> ...
> [from-pstn]
> exten => s,1,Dial(SIP/100,10)
> same => n,VoiceMail(100,u)
>
>
>
>
> root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
> ...
> ...
> ...
> ;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
> ...
> ...
> ...
>
> What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.
>
> I did those modifications in the file /etc/asterisk/chan_dahdi.conf
without improuvement ( After restarting Asterisk)
>
> [channels]
> cidstart=ring
> immediate=yes
> faxdetect=no
> usecallerid=no
>
>
> Here is the debug from Asterisk console
>
> *CLI> -- Starting simple switch on 'DAHDI/1-1'
> -- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new
stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/100
> -- SIP/100-0001 is ringing
> == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
--
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[asterisk-users] Delay before ringing from PSTN`s call
Hi
I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
I configured incoming calls from pstn to ring my SIP phone (extension : 100)
cat extensions.conf
...
[from-pstn]
exten => s,1,Dial(SIP/100,10)
same => n,VoiceMail(100,u)
root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
...
...
...
;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default
...
...
...
What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.
I did those modifications in the file /etc/asterisk/chan_dahdi.conf without
improuvement ( After restarting Asterisk)
[channels]
cidstart=ring
immediate=yes
faxdetect=no
usecallerid=no
Here is the debug from Asterisk console
*CLI> -- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new
stack
== Using SIP RTP CoS mark 5
-- Called SIP/100
-- SIP/100-0001 is ringing
== Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
--
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Re: [asterisk-users] RTP stream when * and Xlite are both behind Nat devices.
Hi rcswebb, I had a problem like yours : Asterisk -NAT - internet - NAT - 3CX phone Without modifiyng Astrisk conf I could start a call from the client but without hearing a sound. The solution for me was to force Asterisk to modify the outgoing udp packet to insert it's public ip and not the private IP behind the NAT . So in your sip.conf I modified : [general] *externip=YouEternalIP* NAT=Yes Hope that'll help :-) Message: 11 Date: Wed, 21 Sep 2011 10:52:08 +0100 From: Richard Webb Subject: [asterisk-users] RTP stream when * and Xlite are both behind Nat devices. To: asterisk-users@lists.digium.com Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp stream to be sent to the public facing address of the softphone? Cheers, -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone numbers and asterisk
I mean the directory of phone numbers is stored within asterisk. So the SIP phone just fetch that list when it starts. -- Message: 3 Date: Sun, 4 Sep 2011 19:47:00 -0400 From: Robert-iPhone Subject: Re: [asterisk-users] Phone numbers and asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <0d19da7a-0766-4ae9-a967-528ccae36...@gmail.com> Content-Type: text/plain; charset="us-ascii" what do you mean? Like speed dial or directory? Sent from my iPhone On Sep 4, 2011, at 6:47 PM, neo haux wrote: > Hi, > > It is possible to save all the phones numbers on asterisk servers instead of doing so manually in each VoIP device ? > > Does SIP take care of such configuration ? > > Thanks -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone numbers and asterisk
Hi, It is possible to save all the phones numbers on asterisk servers instead of doing so manually in each VoIP device ? Does SIP take care of such configuration ? Thanks -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk+internal phones+recorded messages
Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
Hi Asterisk lovers, Thank you very much Warren and Tzafrir ! I resolved the issu by installing openssl-dev. Date: Tue, 2 Aug 2011 15:40:17 -0500 From: Warren Selby Subject: Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice) To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <7648899e-b872-4d34-91a0-e67c73061...@selbytech.com> Content-Type: text/plain; charset=utf-8 Install OpenSSL-devel (or whatever the equivalent ubuntu package is called) and then recompile / reinstall and test it again. Thanks, --Warren Selby, dCAP On Aug 2, 2011, at 12:06 PM, neo haux wrote: > Hi, > > I?ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy) > I also compiled iksemel (v1.4) with the option 2./configure > --with-libgnutls-prefix=/usr" > As explained in this link (to avoid compilation error ) > http://code.google.com/p/iksemel/issues/detail?id=29#c3 > > I configured jabber.conf and gtalk.conf as explained in > wiki.asterisk.org, but I have this error when starting : > asterisk -c -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
Hi, I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy) I also compiled iksemel (v1.4) with the option 2./configure --with-libgnutls-prefix=/usr" As explained in this link (to avoid compilation error ) http://code.google.com/p/iksemel/issues/detail?id=29#c3 I configured jabber.conf and gtalk.conf as explained in wiki.asterisk.org, but I have this error when starting : asterisk -c ... ... JABBER: asterisk INCOMING: http://etherx.jabber.org/streams"; xmlns="jabber:client">X-GOOGLE-TOKENX-OAUTH2 [Aug 2 12:57:40] ERROR[31479]: res_jabber.c:1610 aji_act_hook: OpenSSL not installed. You need to install OpenSSL on this system, or disable the TLS option in your configuration file [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:1391 aji_recv: Parsing failure: Hook returned an error. [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:2742 aji_recv_loop: JABBER: Got hook event. [Aug 2 12:57:56] WARNING[31479]: res_jabber.c:2753 aji_recv_loop: JABBER: socket read error But I have already openssl : root@mohamed-desktop:/usr/src# dpkg -l |grep -i openssl ii libcurl3 7.21.0-1ubuntu1.1 Multi-protocol file transfer library (OpenSSL) ii libxmlsec1-openss 1.2.14-1+squeeze1build0.10.10.1 Openssl engine for the XML security library ii openssl 0.9.8o-1ubuntu4.4 Secure Socket Layer (SSL) binary and related cryptographic tools ii python-op 0.10-1 Python wrapper around the OpenSSL library ii ssl-cert 1.0.26 simple debconf wrapper for OpenSSL Have you any idea where is the problem ? NB: I didn´t have that problem with asterisk 1.6 Thx -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
