[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] version compatible with centos 5.7 (2.6.18-308.8.2.el5)
dear guys plz tell me which version of asterisk is compatible with centos 5.7 (2.6.18-308.8.2.el5). and which is the latest version. regards neo -- _ -- 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] freepbx asterisk
dear i am a neewbi for asterisk, plz tell me or if any link is there where i can understand how asterisk, freepbx, web-meetme, dahdi all these tools works and how they are related. plz help me. 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] 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 -- _ -- 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] 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 -- _ -- 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] 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' > -- _ -- 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] 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' -- _ -- 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] 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
Re: [asterisk-users] Asterisk crashes : Failed to start PBX
Hi, I have already setup to rotate logs hourly & Debug level is 3. Is there any other possibility of crash? Thanks in advance!! -- Regards, Voipexpert From: Giorgio Incantalupo To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Fri, November 20, 2009 10:06:06 PM Subject: Re: [asterisk-users] Asterisk crashes : Failed to start PBX Hi Neo, have you checked your log files? It sometimes happened to me that Asterisk crashed without a reason. I discovered my logrotate didn't make its dirty work so I had huge log files. I lowered Asterisk log level and forced logrotate to work and now I have no more crashes. Hope it may help. :) Giorgio. Neo Anderson wrote: > Hello, > > I am using Asterisk 1.4.24.1 version in production. > OS is Centos 5.3 64 bit & RAM is 8 GB. > I am facing crash in asterisk approx each 12 hour. > When it crashes I see below lines in asterisk logs. > [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread > [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( > I debugged asterisk source code in details & I found that it happens > because it can not allocate memory to create thread. > > Another thing is, when I check coredump using gdb, it's not showing > any debug symbols. > > Would you please let me know how to prevent or resolve this? > > Thanks in advance!! > > -- > Regards, > voipexpert > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crashes : Failed to start PBX
Hello, I am using Asterisk 1.4.24.1 version in production. OS is Centos 5.3 64 bit & RAM is 8 GB. I am facing crash in asterisk approx each 12 hour. When it crashes I see below linesin asterisk logs. [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( I debugged asterisk source code in details & I found that it happens because it can not allocate memory to create thread. Another thing is, when I check coredump using gdb, it's not showing any debug symbols. Would you please let me know how to prevent or resolve this? Thanks in advance!! -- Regards, voipexpert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authentication Problem
Hello everbody, I am having problems is Database version and Real time version of Asterisk. Users are connecting with no problem, they gets authenticate and its working fine, but after 2-3 minutes, registration with the same user comes and it gets failed to authenticate. dial tone gone, users unable to call, but this behaviour not remains for the all users for all the time. most of the time they are able to call, its totally wiered to me. any ideas ? -Neo p.s i m using different kinds of clients xlite xpro cisco ata dta ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Alcatel PBX
hey Torsten, i tried overlapdial=yes and guess what ? IT WORKED i owe you one. thanks friend. -Neo >Hello, >On Fri, 19 Nov 2004, pbx wrote: > [EMAIL PROTECTED] wrote: > > >i have the following scnario. > > > >1. Alcatel PBX with e1 module > >2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1 > >connected to alcatel pbx. > > > >i m having problem in outgoing from alcatel. > >incoming from pstn -> asterisk -> alcatel working fine, but outgoing from > >alcatel -> asterisk -> pstn or any sip extensions not working. it hangs up the > >line as soon as i answer the call. i have generated dialtone via playtones but > >it has also issue. > > > >when i connect pstn e1 line directly to altacel e1 module, it works fine, but > >behind asterisk it hangups. > > > >any body have good idea ? Did you set overlapdial=yes in your zapata.conf? > > > >further details can be provided if u need more. > > > >regards. > >-Neo > > > > > > > > > >This message was sent using IMP, the Internet Messaging Program. > > > >___ > >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 > > > > > > > It might be a good Idea, to at least send your config file zaptel, > zapata,and extension > and the message that you got on the console when the problem occurs > ... Regards, > Jack > > ___ > 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 > This message was sent using IMP, the Internet Messaging Program. ___ 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
[Asterisk-Users] Alcatel PBX
Dear Users, i have the following scnario. 1. Alcatel PBX with e1 module 2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1 connected to alcatel pbx. i m having problem in outgoing from alcatel. incoming from pstn -> asterisk -> alcatel working fine, but outgoing from alcatel -> asterisk -> pstn or any sip extensions not working. it hangs up the line as soon as i answer the call. i have generated dialtone via playtones but it has also issue. when i connect pstn e1 line directly to altacel e1 module, it works fine, but behind asterisk it hangups. any body have good idea ? further details can be provided if u need more. regards. -Neo This message was sent using IMP, the Internet Messaging Program. ___ 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
[Asterisk-Users] why ata stop working after 10 mins after registering from mysql
hello, i m using sip.conf and extension.conf from mysqldb my problem is after 10 mins of registration ata stops working, it works fine for outgoing but fail for incoming. any help ? -neo ___ 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
[Asterisk-Users] Voange with asterisk settings
hello, > here comes the settings. > sip.conf > register => username:[EMAIL PROTECTED]:5061/1010 > [vonage] > type=friend > username=yourusername > secret=yourpw > host=sphone.vopr.vonage.net > port=5061 > disallow=all > allow=ulaw > maxexpirey=15 > dtmfmode=inband > fromuser=yourusername > fromdomain=sphone.vopr.vonage.net > canreinvite=no > nat=yes > > in extensions.conf > > exten => _1.,1,Dial(SIP/[EMAIL PROTECTED],1000,tr) > for inbound in exten.conf > exten => _yourusername,1,whateveryouwanttodo enjoy guys :) let me know if u feel prob ___ 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
[Asterisk-Users] Vonage working with asterisk
atlast after working of 7 hours i got voange soft account working on asterisk. ___ 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
[Asterisk-Users] warning message (sound card) - when I run asterisk!!!
All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) My sound card information: Vendor : Intel Corp. Model : 82801CA/CAM AC'97 Audio Controller Module : i810_audio After running 'dial' command under the asterisk prompt, I got the following message without any sound. *CLI> -- Executing Wait("OSS/dsp", "1") in new stack -- Executing Answer("OSS/dsp", "") in new stack << Console call has been answered >> -- Executing DigitTimeout("OSS/dsp", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("OSS/dsp", "10") in new stack -- Set Response Timeout to 10 -- Executing BackGround("OSS/dsp", "demo-congrats") in new stack May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567 oss_write: Unable to set device to input mode May 26 00:40:55 WARNING[-1221268560]: file.c:537 ast_readaudio_callback: Failed to write frame -- Playing 'demo-congrats' (language 'en') == Spawn extension (local, s, 5) exited non-zero on 'OSS/dsp' Is there anyone can give me any hints or help? Thanks, Neo
[Asterisk-Users] warning message (sound card) - when I run asterisk!!!
All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) My sound card information: Vendor : Intel Corp. Model : 82801CA/CAM AC'97 Audio Controller Module : i810_audio After running 'dial' command under the asterisk prompt, I got the following message without any sound. *CLI> -- Executing Wait("OSS/dsp", "1") in new stack -- Executing Answer("OSS/dsp", "") in new stack << Console call has been answered >> -- Executing DigitTimeout("OSS/dsp", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("OSS/dsp", "10") in new stack -- Set Response Timeout to 10 -- Executing BackGround("OSS/dsp", "demo-congrats") in new stack May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567 oss_write: Unable to set device to input mode May 26 00:40:55 WARNING[-1221268560]: file.c:537 ast_readaudio_callback: Failed to write frame -- Playing 'demo-congrats' (language 'en') == Spawn extension (local, s, 5) exited non-zero on 'OSS/dsp' Is there anyone can give me any hints or help? Thanks, Neo
[Asterisk-Users] vonage sip url
Hello List, anybody knows the sip url of vonage ??? like [EMAIL PROTECTED] ?? regards. -Neo ___ 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
[Asterisk-Users] Zap Outgoing
Hello All, i m having busy signal when i dial any number, while incoming on zap is working fine and its transfering to my soft phone. some time back outgoing was working ok but now i dont know what i messed up. any idea ? it gives busy signal after Zap/25-1 answered SIP/300 -Neo = Spawn extension (voicepulse-incoming, s, 1) exited non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' -- Executing Dial("SIP/3000-2e72", "Zap/25/18005558355") in new stack -- Called 25/18005558355 -- Zap/25-1 answered SIP/3000-2e72 ___ 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
[Asterisk-Users] Incoming on Zap
Hello Everybody, I have 1 X100P (FXO) card attached to my *, also there is a line connected to it, but when i dial that number it is not forwarding to anywhere, just * recognizes that the call is coming. I want incoming call to be forward on my x-lite extension lets say 2000, can anybody tell me the settings of extensions.conf and other conf files. thanks and best regards. -Neo ___ 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
[Asterisk-Users] DLINK DPH-70 with asterisk
Hello everybody, i have DLINK DPH-70 Phone, does anybody know if it works with asterisk ? i have g729 codec installed on my asterisk server which the phone supports, and it gets authenticate with asterisk but when i make a call it says maximum tries reaches for dialing.. regards. -neo ___ 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