[asterisk-users] Forwarding inbound mobiles
We have a need for up to a dozen UK mobile numbers to be forwarded to a UK landline. I know that I can just forward them, but was wondering if anyone knew of any deals / contracts with a UK mobile operator that would lessen the cost. At the moment we are looking at going with Vodafone . Thanks 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] Getting calee audio in Asterisk (real time)
Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio - audio in to jack, audio out from jack - current Asterisk application Outgoing call audio - current Asterisk application However, I need vica-versa: Incoming call audio - current Asterisk application Outgoing call audio - Audio from jack, Audio into Jack - current Asterisk application or at least Incoming call audio - current Asterisk application Audio to jack - current Asterisk application Outgoing call audio - current Asterisk application Any idea how I could accomplish this? Regards Motiejus Jakštys -- _ -- 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] BAD ROUND TIME FOR ANSWEREDTIME
Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME for me. I want that when ANSWEREDTIME =~ 5.6 become 6 and if ANSWEREDTIME= 10.3 become 10 because, now, if ANSWEREDTIME =~ 15.9, it become 15! it isn't correct I could manipulate the app_dial.c to have my own result. But do you think that my idea is correct because, If a call is 15.99 Sec it become 15Sec. For a provider (0.99 * n) become a lot of seconds Thank you -- Francois attachment: francois.vcf smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Getting calee audio in Asterisk (real time)
Update: I thought this may be the solution: *CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on (For 1.6.2 it's *dialplan*set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on) Source: voip-info.org%20http://www.voip-info.org/wiki/view/Asterisk+cmd+jack The command opens two jack ports: Channel:input and channel:output. At once command is executed, sound on the caller is gone. Question: what should this CLI command do in reality? Is it a bug or expected behaviour? Then I connect those two ports hoping it will return the sound to the caller: jack_connect SIP/PBX2-000d:output SIP/PBX2-000d:input Then the calee hears garbled sound. Sample of all process is herehttp://www.megaupload.com/?d=10LN8QRH. It is recorded by MixMonitor on the machine where jack takes process. Asterisk 1.6.2.6 (upgrading/downgrading/patching is not a problem). Waiting for your suggestions... Maybe I can do this in totally different approach? Regards Motiejus Jakštys http://m.jakstys.lt/ 2010/5/5 Motiejus Jakštys desired@gmail.com Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio - audio in to jack, audio out from jack - current Asterisk application Outgoing call audio - current Asterisk application However, I need vica-versa: Incoming call audio - current Asterisk application Outgoing call audio - Audio from jack, Audio into Jack - current Asterisk application or at least Incoming call audio - current Asterisk application Audio to jack - current Asterisk application Outgoing call audio - current Asterisk application Any idea how I could accomplish this? -- _ -- 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] Confirm answering a call
Hello, I am working on getting the following to work and I couldn't find it in the documentation I did read. Where should I look or does someone have an example how I can do it? Current situation: Incoming call - 3 SIP phones + 2 mobile phones ring - if mobile phone goes to voicemail the call is answered by that voicemail (if a phone is in use for another call the call directly goes to that voicemail) Situation I want: Incoming call - 3 SIP + 2 mobile phones ring - if the call is answered by a mobile phone the person picking up the call needs to press 1 (or another key on the phone) to answer the phone, if that key is not pressed all phones keep ringing as being it an unanswered call If it is also required to press the key on the SIP phones than that is acceptable. Is it possible? Where should I look? I know some systems use it. Regards, Mark -- _ -- 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] CDR to MS-SQL via ODBC issue
Hi guys, Having issue with getting CDR to write to MS-SQL via ODBC. cdr_odbc: Connected to freetds-connector cdr_odbc: Error in PREPARE -1 cdr_odbc: Query FAILED Call not logged! == Spawn extension (cisco, ##, 2) exited non-zero on 'IAX2/ast-507 Isql test: [...@ asterisk]# isql freetds-connector XXX Y +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same database / freetds, but I cannot write CDRs. Any ideas would be really appreciated. Thanks, Neeraj -- _ -- 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] Confirm answering a call
Mark Scholten wrote: Situation I want: Incoming call - 3 SIP + 2 mobile phones ring - if the call is answered by a mobile phone the person picking up the call needs to press 1 (or another http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] HDLC Receiver overrun on Wildcard TE410P
and it happened again, I've attached kernel logs from dahdi restart on paste-bin http://pastebin.com/drg3WD20 to fix this problem, I have to: stop dahdi and asterisk - /etc/init.d/dahdi stop remove all dahdi modules - rmmod wct4xxp, dahdi_echocan_mg2, dahdi load modules - modprobe dahdi, modprobe wct4xxp start asterisk after that it works like a charm till next time... any idea what's happening and what else to check ? W dniu 30 kwietnia 2010 13:20 użytkownik Łukasz Krzyżak lukasz.krzy...@nao-team.eu napisał: Hello I've got small PBX (30 simultaneous connections) based on asterisk (1.6.2.6), which uses Stargate 2N ISDN to GSM gate. It runs ok for day or two, but then I get: dahdi: HDLC Receiver overrun on channel TE4/0/1/16 (master=TE4/0/1/16) in my kernel logs, in asterisk i get: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active (span 3 is not connected to gateway for now) and I can't make any calls. My dahdi-channels.conf: ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_net channel = 1-15,17-31 context = default group = 63 /etc/dahdi/system.conf: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,0,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 /proc/interrupts: CPU0 CPU1 CPU2 CPU3 0: 462 313 432 0 IO-APIC-edge timer 1: 3 5 5 3 IO-APIC-edge i8042 8: 32 31 32 34 IO-APIC-edge rtc0 9: 0 0 0 0 IO-APIC-fasteoi acpi 12: 28 27 29 30 IO-APIC-edge i8042 14: 3 3 1 0 IO-APIC-edge ata_piix 15: 0 0 0 0 IO-APIC-edge ata_piix 16: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb2, uhci_hcd:usb5 18: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb4 19: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb3 23: 351 350 200 188 IO-APIC-fasteoi ehci_hcd:usb1 25: 0 0 0 1 IO-APIC-fasteoi 26: 2151 2083 104408008 1985 IO-APIC-fasteoi eth1 51: 1089962 1089895 2940 2910 IO-APIC-fasteoi cciss0 78: 485362 485513 485473 320148527 IO-APIC-fasteoi wct4xxp NMI: 0 0 0 0 Non-maskable interrupts LOC: 30245307 31461472 20982472 23492748 Local timer interrupts SPU: 0 0 0 0 Spurious interrupts CNT: 0 0 0 0 Performance counter interrupts PND: 0 0 0 0 Performance pending work RES: 436674 440302 2195268 1451020 Rescheduling interrupts CAL: 169 265 203 251 Function call interrupts TLB: 43920 44257 50177 52884 TLB shootdowns TRM: 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 Machine check exceptions MCP: 1073 1073 1073 1073 Machine check polls ERR: 3 MIS: 0 I manually set irq affinity - eth1 to CPU2, digium card to CPU3, rest of common interrupts to CPU0 and CPU1 PBX runs on HP ProLiant DL380 G5 server, OS is Gentoo Linux with 2.6.31 kernel. Other software versions: asterisk - 1.6.2.6 libpri - 1.4.10.2 dahdi - 2.2.0.2 any idea what could be the problem / what should I check to diagnose it ? Luke -- _ -- 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] SIP - SIP over PBX no audio when canreinvite=no
Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware phone interface. The Alcatel PBX is connected to the public phone net, and is configured to forward all calls to a certain number to Asterisk. Also when Asterisk dials out, that number is correctly transmitted by the PBX. Asterisk's job is to implement a special highly dynamic call routing, controlled by a script. I tested all functionality I need first with simple Softphones from my work PC. Everything I needed worked fine. Now connected to the PBX it works too, but in certain situations, I simply get no audio at all. Call setup and dynamically calling the correct recipient works fine, but if the callee picks up the phone, there simply is only silence on the line. More precisely, I have the following situation: I call a number (with my desktop phone), the number is picked up by the Alcatel PBX and is calling Asterisk via SIP on a specific extension. Asterisk determines the target, initiates a call via SIP out over the Alcatel and the other phone rings (say my mobile). I can pick it up and the call is connected. Now, if I have canreinvite=no, meaning the connection goes like this: Desk Phone - PSTN - Alcatel PBX - SIP - Asterisk and Asterisk - SIP - Alcatel PBX - PSTN - mobile then I hear nothing. There is only silence. Talking with the Alcatel PBX people, they can tell me that their SIP equipment is allocating codec and compressor resources so the media path is open. I can also confirm that there is RTP network traffic passing to and from Asterisk. But there is only silence. If I change it so, that either source or destination of the call is not going through the PBX but to one of my Softphones registered at Asterisk, then it works fine. The Softphone can receive or initiate the call and there is audio between the two. If I set canreinvite=yes and I have set directrtpsetup=yes to say Asterisk I want it to shortcut anyway, since I'm only interested in the call setup and not really in the actual audio/media data, then the above scenario does work. Desktop phone to mobile phone both via Alcatel PBX works fine, except that I don't get the call disconnected when one side hangs up (seems simply to keep the line open with silence from then on). However while that scenario works, call origination then fails. If I perform a call origination, then the first phone rings, and if picked up, the second phone rings exactly once (or actually, it feels like a fraction of one ring) and then the first phone gets a NO ANSWER/BUSY response right ahead. If I remove once again the canreinvite=yes then the second phone rings normally on call origination and can be picked up, but again, I am having no audio, only silence. So in short, with canreinvite=yes, everything through Asterisk, call forward and call origination works but no audio canreinvite=no, call forward works (but no hangup detection), call origination fails with the called member only receiving one single ring. I really can't find any hints to this, but I think it must be a simple configuration issue on my end. I can provide configuration snippets, but I think the issue is something basic that if someone knows what I am doing wrong, can immediately point me to the answer. Otherwise, here's the sip.conf part where the connection is defined to the PBX: [pbx] type=peer secret=something defaultuser=something fromuser=7889 ; extension we are called with host=10.64.x.y ; IP of PBX sip gateway fromdomain=172.29.x.y ; our IP. canreinvite=yes context=pbxIN direct call forward out again (the scenario above) extensions.conf: exten = 7889,1,Dial(SIP/pbx/0phonenumber) If more debug / config information is required, I'll be happy to provide that. Thanks in advance Rene -- _ -- 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] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
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[asterisk-users] AMR codec for Asterisk 1.6.1.X
Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- 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] Hash Dial Pattern Problems
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
_
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Re: [asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!
Thank you Danny, but it says in the link that it's an iptables issue, though
i allowed everything on this network interface and even stopped iptables but
still i have this issue.
2010/5/4 Danny Nicholas da...@debsinc.com
See if this helps
http://www.voipuser.org/forum_topic_3921.html
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Tuesday, May 04, 2010 11:35 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Code in extensions.conf to leave a voice
mailin another PBX ?!
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring
phone for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of the
macro just says “goto dialstatus” with no Dial statement.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, April 29, 2010 2:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1)
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1]
VoiceMail(IAX2/pbx2-15464, u8029) in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
Failed to write frame*
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on
'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
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--
Abdullah
--
Abdullah
--
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--
Abdullah
--
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Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because
my other 2 1.4.30 boxes wouldn’t talk to it properly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:23 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable.
There is no activity in the CLI. Other patterns for the trunk work just
fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
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Re: [asterisk-users] Hash Dial Pattern Problems
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
_
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[asterisk-users] OT: NAT in SPA922
Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way, version is 5.1.15(a). I'll appreciate links for downloading new firmware) Thanks in advance, Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!
This is a little over my head, but the message indicates that you don't have
a fully authorized connection. Can you post the iax.conf snippets relevant
to the call?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Wednesday, May 05, 2010 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Code in extensions.conf to leave a voicemailin
another PBX ?!
Thank you Danny, but it says in the link that it's an iptables issue, though
i allowed everything on this network interface and even stopped iptables but
still i have this issue.
2010/5/4 Danny Nicholas da...@debsinc.com
See if this helps
http://www.voipuser.org/forum_topic_3921.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Tuesday, May 04, 2010 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Code in extensions.conf to leave a voice
mailin another PBX ?!
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring phone
for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of the
macro just says goto dialstatus with no Dial statement.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Code in extensions.conf to leave a voice mail in
another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1) in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail(IAX2/pbx2-15464,
u8029) in new stack
[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
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--
Abdullah
--
Abdullah
--
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--
Abdullah
--
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[asterisk-users] res_config_mysql - maximum field length for appdata
Hello list, as I am trying to write a complex macro for my users i have the problem, that the appdata field in the extensions table is to small for all my macro parameters. I am using the DB definition from http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions so appdata is limited to 128 characters. And I would prefer some more ;) After looking around in res_config_mysql.c I did not find any design based limitation because to me the relevant data structures seems to be allocated dynamically. But as I am not a skilled C-hacker, it will be great if someone could tell me if there will be unwanted side effects after changing the fieldlength to something like 200 characters or more... btw: is there another (maybe more official?!) source for the database definition than voip-info.org? I did not find anything in the addons directory... Regards, Sebastian -- Sebastian Denz sebastian.d...@gonicus.de (System Engineer) * GONICUS GmbH * Zentrale * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 270 * http://www.GONICUS.de *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 -- _ -- 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] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn’t talk to it properly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:23 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable.
There is no activity in the CLI. Other patterns for the trunk work just
fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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To UNSUBSCRIBE or update options visit:
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_
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Re: [asterisk-users] Hash Dial Pattern Problems
Ok - you have to be getting something or you wouldn't get that message. You
are looking at CLI on the 1.2 or 1.4 box? If you're looking at the 1.4 side,
you won't see anything until a connection is made (although you should see
some kind of credential reject or something??)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_
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Re: [asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!
Hi Guys,
first of all, thanks Danny for your support trying to help is a big help
itself.
so the thing is:
from pbx1 to pbx2 which was able to leave VM, it was set up like this:
exten = 8021,1,Dial(IAX2/pbx2/${EXTEN},30,tTWwr)
but from pbx2 to pbx1 which was not able to leave VM, it was setup like
this:
exten = 8093,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
that seems to me suly, but though i wnet ahead and modified the only
difference which is the ring time from 20 to 30, and IT WORKED!!!
i wasted some time going over values, and it seems like it's working for 21
but not for 20, maybe a pro can give us precise explanation, but at least i
can leave a VM now :)!
2010/5/5 Danny Nicholas da...@debsinc.com
This is a little over my head, but the message indicates that you don’t
have a fully authorized connection. Can you post the iax.conf snippets
relevant to the call?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Wednesday, May 05, 2010 8:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Code in extensions.conf to leave a
voicemailin another PBX ?!
Thank you Danny, but it says in the link that it's an iptables issue,
though i allowed everything on this network interface and even stopped
iptables but still i have this issue.
2010/5/4 Danny Nicholas da...@debsinc.com
See if this helps
http://www.voipuser.org/forum_topic_3921.html
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Tuesday, May 04, 2010 11:35 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Code in extensions.conf to leave a voice
mailin another PBX ?!
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring
phone for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of the
macro just says “goto dialstatus” with no Dial statement.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, April 29, 2010 2:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1)
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1]
VoiceMail(IAX2/pbx2-15464, u8029) in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
Failed to write frame*
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on
'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Re: [asterisk-users] OT: NAT in SPA922
On 5 May 2010, at 14:39, Sebastian Milioto wrote: However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way, version is 5.1.15(a). I'll appreciate links for downloading new firmware) It's a phone not a router. It doesn't do nat. You can get new firmware from www.cisco.com (believe free CCO login will get you the SMB stuff). The 'My Cisco Community' forums are also good. Has real Cisco people who appear to know their stuff. S -- _ -- 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] Hash Dial Pattern Problems
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device
3000
May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote:
Ok – you have to be getting something or you wouldn’t get that message.
You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4
side, you won’t see anything until a connection is made (although you should
see some kind of credential reject or something??)
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 9:31 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn’t talk to it properly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:23 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable.
There is no activity in the CLI. Other patterns for the trunk work just
fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
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Re: [asterisk-users] CDR to MS-SQL via ODBC issue
On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote: I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same database / freetds, but I cannot write CDRs. Are you writing to the database with func_odbc, or just reading? My gut says that you need to check your permissions on the database to ensure that you're allowed to write to the CDR table. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] OT: NAT in SPA922
However, when I connect a PC to that port, SPA922 works as bridge. Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike the SPA2102, etc). I think the 5.1 series is the latest firmware for the 922; the the 942, there is 6.1.5a. Luki -- _ -- 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] What is billsec in CDR?
In my system (Asterisk 1.4.30) I found that if I have some playback() or saydigit() before dial(), the billsec in CDR count all the time includes the playback time. For example, if I dial a number, listen the playback, then just hangup before the call get answered, the CDR show me the time spent doing the playback in the billsec. CDR has two fields - duration and billsec. My understand is the billsec start count when a call is answered. Am I right? or Am I missing something here. -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- 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] AMR codec for Asterisk 1.6.1.X
== Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Probably shouldn't be listing it as unknown Have you tried using that AMR codec beyond commands in the asterisk cli? Did the patch apply cleanly? On Wed, May 5, 2010 at 6:21 AM, Andrea Cristofanini andrea.cristofan...@zerozero39.it wrote: Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2 2 2 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 1 2 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 - 2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2 - 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 2 2 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 1 1 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - - - - - - - - - - - - ilbc - - - - - - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 12001 12001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8) (0x100) audio g729 (G.729A) 512 (1 9) (0x200) audio speex (SpeeX) 1024 (1 10) (0x400) audio ilbc (iLBC) 2048 (1 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] AMR codec for Asterisk 1.6.1.X
It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- 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] VoIP Termination in Japan
Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- _ -- 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] Hash Dial Pattern Problems
Ok. I'm confused. I was interpreting what you wrote to say that you are
doing this:
1. pick up sip phone attached to pbx1 (1.2 box)
2. dial #5551212
3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
4. 1.4 box should fall into _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device
3000
May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote:
Ok - you have to be getting something or you wouldn't get that message. You
are looking at CLI on the 1.2 or 1.4 box? If you're looking at the 1.4 side,
you won't see anything until a connection is made (although you should see
some kind of credential reject or something??)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
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Re: [asterisk-users] Registering a Cisco 7965 on 1.4.26
Maybe a rtp.conf problem - normal values are 1-2. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Wednesday, May 05, 2010 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco 7965 on 1.4.26 Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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 -- _ -- 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] Registering a Cisco 7965 on 1.4.26
Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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] run script after completed
DeadAGI is deprecated in Asterisk 1.6.x ! 2010/4/9 Danny Nicholas da...@debsinc.com Do the call in a context and have the context run the script as a DeadAGI. [call_and_do] - exten = s,1,Dial… - exten = h,1,Deadagi(…) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir *Sent:* Friday, April 09, 2010 7:34 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] run script after completed Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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 -- _ -- 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] Registering a Cisco 7965 on 1.4.26
On Wed, May 5, 2010 at 10:16 AM, Danny Nicholas da...@debsinc.com wrote: Maybe a rtp.conf problem - normal values are 1-2. I haven't even gotten to the RTP stage, it won't even register on the SIP side because responses are being sent back to the wrong SIP signaling port. -- James -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Wednesday, May 05, 2010 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco 7965 on 1.4.26 Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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] run script after completed
Regular AGI with SIGHUP detection? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mickael Monsieur Sent: Wednesday, May 05, 2010 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] run script after completed DeadAGI is deprecated in Asterisk 1.6.x ! 2010/4/9 Danny Nicholas da...@debsinc.com Do the call in a context and have the context run the script as a DeadAGI. [call_and_do] - exten = s,1,Dial. - exten = h,1,Deadagi(.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir Sent: Friday, April 09, 2010 7:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] run script after completed Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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 -- _ -- 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] VoIP Termination in Japan
probe with this: www.siptraffic.com Our company have a lot of experience with this company through a lot of routes and simply they are the best we know in quality/price rate. They ask you for 200$ initially, but they work perfectly! The only problem is that they never give a CID number. If you don't have any problem with this, they give you really good service. Regards, Miguel Amez 2010/5/5 Adrian Marsh adrian.ma...@ubiquisys.com Anyone have any experience with a Japanese local VoIP termination supplier? I’ve emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- _ -- 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 -- _ -- 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] What is billsec in CDR?
On 05-05-2010 18:00, Jian Gao wrote: In my system (Asterisk 1.4.30) I found that if I have some playback() or saydigit() before dial(), the billsec in CDR count all the time includes the playback time. For example, if I dial a number, listen the playback, then just hangup before the call get answered, the CDR show me the time spent doing the playback in the billsec. apps like playback do an implicit answer and this fires up the billsec counter. -- _ -- 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] Channels In Use
Hi List, If we have a scenario where a customer is using a telephone and their WAN link goes down for example the channel in asterisk stays marked as in use and this affects the subscribe also. *CLI core show channels Channel Location State Application(Data) SIP/107-Customer-09abc (None) Up AppDial((Outgoing Line)) SIP/101-Customer-09abe s...@macro-tl-userexten Up Dial(SIP/107-Customer|20|rtT) The only way to get rid of this active channel I can find is to restart now. If I restart when convenient asterisk will never restart because it thinks there is a channel in use. Were running pure SIP, no PSTN/ISDN trunks on Asterisk 1.4.22 on Debian 5.03. Are there any CLI commands to free this up or any other ways without having to restart asterisk. Regards, Stephen. -- _ -- 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] IAX2 Auto-congesting call due to slow response
Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: -- -- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 04825 DCall: 0 [10.20.0.201:41764] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw) CALLING NUMBER : 2000 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: athens_user LANGUAGE: en USERNAME: wtgpl FORMAT : 4 CAPABILITY : 4 ADSICPE : 2 DATE TIME : 2010-05-04 18:48:48 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 04825 DCall: 0 [10.20.0.201:41764] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw) CALLING NUMBER : 2000 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: athens_user LANGUAGE: en USERNAME: wtgpl FORMAT : 4 CAPABILITY : 4 ADSICPE : 2 DATE TIME : 2010-05-04 18:48:48 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 02002ms SCall: 0 DCall: 04825 [10.20.0.201:41764] FORMAT : 4 -- -- Asterisk doesn't respond to the last message, and I can't understand why. In asterisk 1.2 it works fine with the following debug: -- -- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 07531 DCall: 0 [10.20.0.201:55767] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw) CALLING NUMBER : 227 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Admin 2 LANGUAGE: en FORMAT : 4 CAPABILITY : 63492 ADSICPE : 2 DATE TIME : 2010-05-04 19:26:02 -- Called wtgpl Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 07531 DCall: 0 [10.20.0.201:55767] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw) CALLING NUMBER : 227 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Admin 2 LANGUAGE: en FORMAT : 4 CAPABILITY : 63492 ADSICPE : 2 DATE TIME : 2010-05-04 19:26:02 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 02007ms SCall: 0 DCall: 07531 [10.20.0.201:55767] FORMAT : 4 -- Call accepted by 10.20.0.201 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK -- -- Thanks in advance, Alex. -- _ -- 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] Transfer calls using ##
I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an extension using ##. 3) Get the voice mail greeting of the individual. 4) Hit 0 for the operator before the greeting completed. 5) Ask the operator to transfer me again using ##. 6) Operator can't transfer and I can hear the pressing of the keys. Why can't I transfer the call the second time around? How can I fix this? The dial statement in your 'o' extension must have the 't' flag. - Noah -- _ -- 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] T38 trunk configuration for relay appears to affect default trunks for voip
Hi list! I have this configuration for sending T38 faxes to my T38 fax termination provider: T38modem -- hylafax -- Asterisk-SIP-Extension -- T38 termination provider -- T.30 termination to PSTN We are experiencing 2 problems with this (if you want configuration files, it won't be a problem, just tell me): 1. T38 termination provider receives faxes at 2400 bpps from our server. This issue could be produced by the bug indicated previously on this list related with a fix that will appear on this week's 1.6.2.8 rc1, I will try with it and tell you. 2. Second problem is what I want to talk about on this mail: We are detecting some extrange behaviour on the VoIP outgoing trunks that worked fine before we installed T38modem and Asterisk's configuration to support it. Calls are received by the people we call to, but they can't hear nothing and we don't receive any kind of signal or tone in the phones. It's more and more extrange if I tell you that if we hang the call and retry a few seconds later, the call is made perfectly. We got some logs when this happened: [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using SIP RTP TOS bits 184 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using SIP RTP CoS mark 5 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using UDPTL TOS bits 184 [Apr 30 09:34:31] VERBOSE[5296] netsock.c: == Using UDPTL CoS mark 5 [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [034635933...@from- internal:1] Macro(SIP/21-0058, user-callerid,SKIPTTL,) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:1] Set(SIP/21-0058, AMPUSER=21) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:2] GotoIf(SIP/21-0058, 0?report) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:3] ExecIf(SIP/21-0058, 1?Set(REALCALLERIDNUM=21)) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:4] Set(SIP/21-0058, AMPUSER=21) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:5] Set(SIP/21-0058, AMPUSERCIDNAME=Aula 11) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:6] GotoIf(SIP/21-0058, 0?report) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:7] Set(SIP/21-0058, AMPUSERCID=21) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:8] Set(SIP/21-0058, CALLERID(all)=Aula 11 21) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:9] GotoIf(SIP/21-0058, 1?continue) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Goto (macro-user-callerid,s,18) [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-user-callerid:18] NoOp(SIP/21-0058, Using CallerID Aula 11 21) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [034635933...@from-internal:2] Set(SIP/21-0058, _NODEST=) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [034635933...@from-internal:3] Macro(SIP/21-0058, record-enable,21,OUT,) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-record-enable:1] GotoIf(SIP/21-0058, 1?check) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Goto (macro-record-enable,s,4) [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-record-enable:4] ExecIf(SIP/21-0058, 0?MacroExit()) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-record-enable:5] GotoIf(SIP/21-0058, 0?Group:OUT) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Goto (macro-record-enable,s,16) [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-record-enable:16] GotoIf(SIP/21-0058, 0?IN) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-record-enable:17] ExecIf(SIP/21-0058, 1?MacroExit()) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [034635933...@from-internal:4] Macro(SIP/21-0058, dialout-trunk,2,034635933565,,) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-dialout-trunk:1] Set(SIP/21-0058, DIAL_TRUNK=2) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/21-0058, 0?sub-pincheck,s,1) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/21-0058, 0?disabletrunk,1) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-dialout-trunk:4] Set(SIP/21-0058, DIAL_NUMBER=034635933565) in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [...@macro-dialout-trunk:5] Set(SIP/21-0058, DIAL_TRUNK_OPTIONS=tr) in new stack [Apr 30 09:34:31] VERBOSE[12649]
Re: [asterisk-users] Hash Dial Pattern Problems
Your interpretation is right ownvery weird problem. The problem is when
i dial #551212 there is absolutely no activity in the CLI. It is almost like
there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote:
Ok. I’m confused. I was interpreting what you wrote to say that you are
doing this:
1. pick up sip phone attached to pbx1 (1.2 box)
2. dial #5551212
3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
4. 1.4 box should fall into _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for
device 3000
May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote:
Ok – you have to be getting something or you wouldn’t get that message.
You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4
side, you won’t see anything until a connection is made (although you should
see some kind of credential reject or something??)
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 9:31 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn’t talk to it properly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:23 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable.
There is no activity in the CLI. Other patterns for the trunk work just
fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
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Re: [asterisk-users] Hash Dial Pattern Problems
From 1.2 CLI, do dialplan show _...@default - this will tell you if your
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
Your interpretation is right ownvery weird problem. The problem is when
i dial #551212 there is absolutely no activity in the CLI. It is almost like
there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote:
Ok. I'm confused. I was interpreting what you wrote to say that you are
doing this:
1. pick up sip phone attached to pbx1 (1.2 box)
2. dial #5551212
3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
4. 1.4 box should fall into _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device
3000
May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote:
Ok - you have to be getting something or you wouldn't get that message. You
are looking at CLI on the 1.2 or 1.4 box? If you're looking at the 1.4 side,
you won't see anything until a connection is made (although you should see
some kind of credential reject or something??)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other patterns for the trunk work just fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
--
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Re: [asterisk-users] Hash Dial Pattern Problems
I set: sip debug peer 3000 (my test extension) and dialed #3643873 Here is the output: -- SIP read from 192.168.1.59:17456: INVITE sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:3...@192.168.1.59:17456 To: #3643873sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 From: Testsip:3...@192.168.2.10 sip%3a3...@192.168.2.10;tag=76126b35 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=3000,realm=asterisk,nonce=6a7a2c99,uri= sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 ,response=7bcf9339c154ef939bd575aeaaef1860,algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 317 v=0 o=- 8 2 IN IP4 192.168.1.59 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.59 t=0 0 m=audio 34194 RTP/AVP 107 0 8 101 a=alt:1 2 : MsRCET/S fNqrHReN 192.168.200.113 34194 a=alt:2 1 : pf8wX3Si UdjtGUj2 192.168.1.59 34194 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 12 lines)--- Using INVITE request as basis request - NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. Sending to 192.168.1.59 : 17456 (non-NAT) Found user '3000' Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.59:34194 Found description format BV32 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for %233643873 in from-internal (domain 192.168.2.10) Reliably Transmitting (no NAT) to 192.168.1.59:17456: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport;received=192.168.1.59 From: Testsip:3...@192.168.2.10 sip%3a3...@192.168.2.10;tag=76126b35 To: #3643873sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 ;tag=as3d020428 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 Content-Length: 0 --- aikphone*CLI -- SIP read from 192.168.1.59:17456: ACK sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport To: #3643873sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 ;tag=as3d020428 From: Hull Barrettsip:3...@192.168.2.10 sip%3a3...@192.168.2.10 ;tag=76126b35 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call 'NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.' On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Ok. I’m confused. I was interpreting what you wrote to say that you are doing this: 1. pick up sip phone attached to pbx1 (1.2 box) 2. dial #5551212 3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box 4. 1.4 box should fall into _XXX and do DAHDI dial? If this is correct, where is the IAX command in your CLI output. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel *Sent:* Wednesday, May 05, 2010 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems I am on the 1.2 box and see nothing with the verbose cranked up. I do see the following when tailing the asterisk full log during the calls: May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0 May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device 3000 May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on 'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote: Ok – you have to be getting something or you wouldn’t get that message. You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4 side, you won’t see anything until a connection is made (although you should see some kind of credential reject or something??) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel *Sent:* Wednesday, May 05, 2010 9:31 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Hash Dial Pattern
Re: [asterisk-users] Registering a Cisco 7965 on 1.4.26
James, I'm assuming your talking SIP here. It's not the Contact header that is important here but the Via header. Responses should be going back to whatever port is specified there. Contact header is used for incoming requests. Via header is used for responses to outgoing requests. 7965: 10.1.1.1:44392 - register (Via: SIP/2.0/UDP 10.1.1.1:5060;branch=) - asterisk asterisk should send a response back to 10.1.1.1:5060 from asterisk cli, run 'sip set debug' and post a copy of the REGISTER and asterisk's response. -David -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of James Lamanna Sent: Wed 5/5/2010 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco 7965 on 1.4.26 Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James -- _ -- 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 -- _ -- 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] working example of t38 fax w/ 1.6.2?
2010/5/4 sean darcy seandar...@gmail.com
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on
the better way.
regards,
Miguel Amez
2010/5/4 sean darcy seandar...@gmail.com mailto:seandar...@gmail.com
Miguel Amez wrote:
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com mailto:seandar...@gmail.com
mailto:seandar...@gmail.com
I can't get a test T.38 fax between 2 1.6.2 machines, using
app
_fax and spandsp pre17 and 20100501. The machines can't seem
to get
connected.
send side extensions.conf:
[fax-tx-test]
exten=s,1,NoOp(Context fax-tx-test)
exten=s,n,SendFAX(${FaxFile}.tif)
exten=s,n,HangUp()
exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR}
FAXMODE:
${FAXMODE})
Channel:SIP/side-sip-fax
Context:fax-tx-test
Extension:s
Priority:1
Set:FaxFile=/var/spool/asterisk/fax/20091113_1455
receive side:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif)
exten = s,n,Hangup()
There's a bunch more stuff at
https://issues.asterisk.org/view.php?id=17105
But does anyone have a setup that Just Works? I'd love to
find a setup
that works for someone else and just copy it.
Thanks,
sean
Yes, I am familiar with Hylafax. But I'm trying to Keep It Simple,
and
just use app_fax. Is it working for anyone? Does anybody have a
simple
working example?
sean
It's the fax module built into 1.6.2.
This module doesn't support T.38 (if 'm not mistaken).
sean
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Re: [asterisk-users] What is billsec in CDR?
ad...@3a.hu wrote:
On 05-05-2010 18:00, Jian Gao wrote:
In my system (Asterisk 1.4.30) I found that if I have some playback() or
saydigit() before dial(), the billsec in CDR count all the time includes
the playback time. For example, if I dial a number, listen the playback,
then just hangup before the call get answered, the CDR show me the time
spent doing the playback in the billsec.
apps like playback do an implicit answer and this fires up the billsec
counter.
OK, here is my dialplan:
exten = _011X.,1,Set(remainMinutes=${DB(timer/bbFreeLDMinute)})
exten = _011X.,n,Playback(this-call-will-end-in)
exten = _011X.,n,SayNumber(${remainMinutes})
exten = _011X.,n,Playback(minutes)
exten = _011X.,n,Set(ms=${MATH(${remainMinutes}*6,int)}) ;convert
minutes to ms
exten = _011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3)) ;set
call limit to ${ms}, warning when 3ms(30 sec) left
Is there any way that Asterisk will record the correct billsec? Or, is
there a different approach?
Thanks for help.
--
Jian Gao
IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com
jian@sjgeophysics.com mailto:jian@sjgeophysics.com
Tel: (604)582-1100
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Re: [asterisk-users] Hash Dial Pattern Problems
It doesnt seem to like the _X. . What is this suppose represent?
Thanks
On Wed, May 5, 2010 at 5:53 PM, Danny Nicholas da...@debsinc.com wrote:
From 1.2 CLI, do “dialplan show _...@default – this will tell you if your
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 4:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
Your interpretation is right ownvery weird problem. The problem is
when i dial #551212 there is absolutely no activity in the CLI. It is almost
like there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote:
Ok. I’m confused. I was interpreting what you wrote to say that you are
doing this:
1. pick up sip phone attached to pbx1 (1.2 box)
2. dial #5551212
3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
4. 1.4 box should fall into _XXX and do DAHDI dial?
If this is correct, where is the IAX command in your CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for
device 3000
May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on
'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found
On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas da...@debsinc.com wrote:
Ok – you have to be getting something or you wouldn’t get that message.
You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4
side, you won’t see anything until a connection is made (although you should
see some kind of credential reject or something??)
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 9:31 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn’t talk to it properly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:23 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Hash Dial Pattern Problems
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable.
There is no activity in the CLI. Other patterns for the trunk work just
fine.
Dial pattern:
#|. or #|NXX
exten = _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r)
exten = _#.,2,Congestion
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated.
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[asterisk-users] Still true: only first peer matched on incoming call?
I've got two 1.6.2 asterisk boxes. I'd like to be able to set up two separate sip connections. But when I try that I get: chan_sip.c:12671 check_auth: username mismatch, have one-sip-peer, digest has another-sip-peer Looking around I found this in a 2007 bug report on version 1.4.4, https://issues.asterisk.org/view.php?id=9678: THis is well known. There is a lot of available documentation out there. Basically: We only match the first peer on the incoming call, which is the last peer in the sip.conf file. Yes, I know it is awkward, but it is the way it works now. Still the case? Or is there some clever way around this? sean -- _ -- 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] What is billsec in CDR?
Hi!
apps like playback do an implicit answer and this fires up the billsec
counter.
OK, here is my dialplan:
exten = _011X.,n,Playback(this-call-will-end-in)
exten =
_011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3))
Is there any way that Asterisk will record the correct billsec? Or, is
there a different approach?
Place a ResetCDR() after your Playback() statement and before Dial().
Philipp
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Re: [asterisk-users] Hash Dial Pattern Problems
Hi! I set: sip debug peer 3000 (my test extension) and dialed #3643873 Your X-Lite softphone actually calls %233643873 and not #3643873. You would need to check the SIP RFCs in order to find out if Asterisk is behaving correctly here by not decoding %23 as #. In the meanwhile you could try to add the extension %233643873 to your dialplan, or find out if you can configure the way X-Lite handles the # within the dialstring. To: #3643873sip:%233643...@192.168.2.10 ... User-Agent: X-Lite release 1104o stamp 56125 (telephone-event) Looking for %233643873 in from-internal (domain ... SIP/2.0 404 Not Found Via: SIP/2.0/UDP Philipp -- _ -- 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] Registering a Cisco 7965 on 1.4.26
On Wed, May 5, 2010 at 5:05 PM, David White david.wh...@watchguard.comwrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of James Lamanna Sent: Wed 5/5/2010 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Registering a Cisco 7965 on 1.4.26 Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to send requests back to the high port number. I tried this on 1.6.0.9 with nat=no and everything works fine. Is this a problem with 1.4.26? Is there a 1.4.x version that works? Or did I manage to screw something up? Thanks. -- James Have you tried testing on the latest version of the 1.4.x branch? I believe 1.4.31 was released a couple days ago. Are your phones on the same LAN as your asterisk server? I only found the port issue to be an issue when I had a phone at a client house or office and they were connecting to a public server. Which version of the SIP firmware are you using on your 7965? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Channels In Use
Are there any CLI commands to free this up or any other ways without having to restart asterisk. Did you try soft hangup channel? Or set an RTP timeout to avoid abandoned channels? Luki -- _ -- 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] CDR to MS-SQL via ODBC issue
--- Message: 10 Date: Wed, 5 May 2010 10:26:34 -0500 From: Tilghman Lesher tles...@digium.com Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 201005051026.34929.tles...@digium.com Content-Type: text/plain; charset=iso-8859-1 On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote: I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same database / freetds, but I cannot write CDRs. Are you writing to the database with func_odbc, or just reading? My gut says that you need to check your permissions on the database to ensure that you're allowed to write to the CDR table. Hi Tilghman, yeah I thought so too at first but then, using the same permissions I'm doing both read writes as well. On the database end, the user is setup as database_owner and has db_read db_write permissions. I got Leif to check this with me last night, we couldn't figure it out. The error that pops up is: cdr_odbc: Connected to asterisk-freetds-connector cdr_odbc: Error in PREPARE -1 cdr_odbc: Query FAILED Call not logged! __ -- _ -- 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] CDR to MS-SQL via ODBC issue
On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote: --- Message: 10 Date: Wed, 5 May 2010 10:26:34 -0500 From: Tilghman Lesher tles...@digium.com Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 201005051026.34929.tles...@digium.com Content-Type: text/plain; charset=iso-8859-1 On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote: I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same database / freetds, but I cannot write CDRs. Are you writing to the database with func_odbc, or just reading? My gut says that you need to check your permissions on the database to ensure that you're allowed to write to the CDR table. Hi Tilghman, yeah I thought so too at first but then, using the same permissions I'm doing both read writes as well. On the database end, the user is setup as database_owner and has db_read db_write permissions. I got Leif to check this with me last night, we couldn't figure it out. The error that pops up is: cdr_odbc: Connected to asterisk-freetds-connector cdr_odbc: Error in PREPARE -1 cdr_odbc: Query FAILED Call not logged! __ Okay, second idea is that you should very carefully examine your CDR table layout and ensure that the columns that you have match EXACTLY what the module expects you to have. If Asterisk expects you to have a column that you don't (or the column type is wrong), that is another reason that the prepare might fail. You might consider using the cdr_adaptive_odbc driver, instead, as it is designed to create the insert based upon the structure of the table. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
