[asterisk-users] Anyone have a reliable T.38 Solution
Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt -- _ -- 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] Speech recognition in asterisk using google voice API
On 01/04/2012 07:51 AM, Bruce B wrote: And with recent version 14.3.2 I get: /usr/local/bin/sox FAIL formats: no handler for file extension `flac' -- speech-recog.agi: /usr/local/bin/sox failed: 512 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0 Regards, On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote: Very interesting. I just tried to get it to work but it complains about sox. Probably you used a different version of sox? *PBX-*CLI /usr/bin/sox: invalid option -- -* */usr/bin/sox: invalid option -- n* */usr/bin/sox: invalid option -- o* */usr/bin/sox: -r must be given a positive integer* * -- speech-recog.agi: /usr/bin/sox failed: 512* I am using: *Package sox-12.18.1-1.el5_5.1.i386 * Thanks, Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris -- _ -- 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] Rami
Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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] Failed to authenticate on INVITE to Anonymous
Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Asterisk won't start - trap invalid opcode
Hi there Happy New Year I have a new install of asterisk 1.8.8.1 on ubuntu server 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux It has a Sangoma A200 card and I thought should be fairly standard but I have a new error when trying to start asterisk and I don't really know where to start Initially asterisk was installed with dahdi from a package but sangoma didn't seem happy. Once I added dahdi from source sangoma wanpipe installed okay, but when I reloaded asterisk it stopped. So I removed all the packages (I believe I have but something could be hanging around) and rebuilt asterisk from source. Same errors. The only errors I can see are limited - I also stopped wan router and dahdi and I still get ~# asterisk -cvv Illegal instruction Which isn't very informative. Kind of a fun challenge but not one I need right now Google hasn't been able to find a similar issue My choices that I can see are: - try another version of asterisk - delete everything and start again (which I thought I have tried but maybe not thorough enough) - earlier version of ubuntu But I would really like to understand what's clashing. The sangoma card details are from lspci 04:03.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: NEC Corporation Device 0700 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 255 (1250ns min, 3750ns max) Interrupt: pin A routed to IRQ 17 Region 0: Memory at dfdf (32-bit, non-prefetchable) [size=64K] Output of dmesg looks fine until it gets to asterisk right at the bottom which gives this [ 13.419761] asterisk[1352] trap invalid opcode ip:5316f3 sp:7fff2db1a0f0 error:0 in asterisk[40+1d6000] dmesg long version [5.051430] WANPIPE(tm) Hardware Support Module 3.5.24.0 (c) 1994-2010 Sangoma Technologies Inc [5.052051] usbcore: registered new interface driver sdlausb [5.063329] dahdi: Telephony Interface Registered on major 196 [5.063334] dahdi: Version: 2.5.0.2 [5.071841] WANPIPE(tm) Interface Support Module 3.5.24.0 (c) 1994-2010 Sangoma Technologies Inc [5.098753] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.24.0 (c) 1994-2010 Sangoma Technologies Inc [5.098759] wanpipe: Probing for WANPIPE hardware. [5.098846] pci :04:03.0: PCI INT A - GSI 17 (level, low) - IRQ 17 [5.110126] wanpipe: AFT-A200-SH PCI FXO/FXS card found (HDLC rev.7), cpu(s) 1, bus #4, slot #3, irq #5 [5.110137] wanpipe: Allocating maximum 1 devices: wanpipe1 - wanpipe1. [5.110606] WANPIPE: TDM Codecs Initialized [5.138361] WANPIPE(tm) Socket API Module 3.5.24.0 (c) 1994-2010 Sangoma Technologies Inc [5.138368] NET: Registered protocol family 25 [5.171801] WANPIPE(tm) WANEC Layer 3.5.24.0 (c) 1995-2006 Sangoma Technologies Inc. [5.171807] wanec_create_dev: Registering Wanpipe ECDEV Device! [5.211707] wanpipe1: Starting WAN Setup [5.211713] [5.211715] Processing WAN device wanpipe1... [5.211719] wanpipe1: Locating: A200/A400/B600/B700/B800 card, CPU A, PciBus=4, PciSlot=3 [5.211728] wanpipe1: Found: A200/A400/B600/B700/B800 card, CPU A, PciBus=4, PciSlot=3, Port=0 [5.211785] wanpipe1: AFT PCI memory at 0xDFDF [5.211788] wanpipe1: IRQ 17 allocated to the AFT PCI card [5.211841] wanpipe1: Starting AFT Analog Hardware Init. [5.211874] wanpipe1: Enabling front end link monitor [5.211879] wanpipe1: Global Chip Configuration: used=1 used_type=1 [5.211906] wanpipe1: Global Front End Configuration! [5.211909] wanpipe1: Configuring FXS/FXO Front End ... [5.424637] wanpipe1: Module 1: Installed -- Auto FXO (AUSTRALIA mode)! [5.624587] wanpipe1: Module 2: Installed -- Auto FXO (AUSTRALIA mode)! [5.824637] wanpipe1: Module 3: Installed -- Auto FXO (AUSTRALIA mode)! [6.025333] wanpipe1: Module 4: Installed -- Auto FXO (AUSTRALIA mode)! [6.225169] wanpipe1: Module 5: Installed -- Auto FXO (AUSTRALIA mode)! [6.425906] wanpipe1: Module 6: Installed -- Auto FXO (AUSTRALIA mode)! [6.624559] wanpipe1: Module 7: Installed -- Auto FXO (AUSTRALIA mode)! [6.824553] wanpipe1: Module 8: Installed -- Auto FXO (AUSTRALIA mode)! [6.824560] wanpipe1: Running post initialization... [6.824563] wanpipe1: Remora config done! [6.824567] wanpipe1: AFT Data Mux Bit Map: 0x01234567 [6.824794] wanpipe1: Front End Interface Ready 0x [6.824805] wanpipe1: Register EC interface wanec1 (usage 1, max ec chans 32)! [6.824811] wanpipe1: Configuring Device :wanpipe1 FrmVr=07 [6.824814] wanpipe1:Global MTU = 1500 [6.824817] wanpipe1:Global MRU = 1500 [6.824819] wanpipe1:Data Mux Map = 0x01234567 [6.824821] wanpipe1:
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On Wednesday 04 January 2012, Duncan Turnbull wrote: Hi there Happy New Year I have a new install of asterisk 1.8.8.1 on ubuntu server 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux It has a Sangoma A200 card and I thought should be fairly standard but I have a new error when trying to start asterisk and I don't really know where to start Initially asterisk was installed with dahdi from a package but sangoma didn't seem happy. Once I added dahdi from source sangoma wanpipe installed okay, but when I reloaded asterisk it stopped. So I removed all the packages (I believe I have but something could be hanging around) and rebuilt asterisk from source. Same errors. The only errors I can see are limited - I also stopped wan router and dahdi and I still get ~# asterisk -cvv Illegal instruction Which isn't very informative. Kind of a fun challenge but not one I need right now Google hasn't been able to find a similar issue For what it's worth, I once tried installing Asterisk on an old VIA C7 box; and it turns out that this processor, while detecting as an i686, doesn't implement the full i686 instruction set -- and Asterisk is trying to use one of the non-implemented instructions. Solution was to re-compile for i586. It's just possible that something similar is going on here -- maybe your processor isn't implementing an instruction that Asterisk or Dahdi is relying on. (It's my understanding that 64-bit processors don't fully implement the 32-bit instructions when in 64-bit mode, but I wouldn't swear to that.) Or maybe it's a library path problem -- something trying to use a 32-bit library instead of a 64-bit one, or vice versa. Try ldd on the binaries. What is your output from `cat /proc/cpuinfo` ? If you have at least two SIP phones and/or an IAX route, try disabling Dahdi, and see if you can persuade Asterisk to run like that. At least that should help track the problem down to one layer (Asterisk or Dahdi). -- AJS Answers come *after* questions. -- _ -- 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] Anyone have a reliable T.38 Solution
I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Wednesday, 4 January 2012 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Anyone have a reliable T.38 Solution Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi checked your debug like. Did you check that your SIP device ir registered with server ? if yes then dial below command from CLI *originate sip/test02 application dial* On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On 4/01/2012, at 11:47 PM, A J Stiles wrote: For what it's worth, I once tried installing Asterisk on an old VIA C7 box; and it turns out that this processor, while detecting as an i686, doesn't implement the full i686 instruction set -- and Asterisk is trying to use one of the non-implemented instructions. Solution was to re-compile for i586. Thanks very much AJ That did appear as one of the few google comments I found but I couldn't figure out whether it applies The outputs are below if you can interpret them, I can see lm in the cpu proc info but don't know how to check for better compatibility It's just possible that something similar is going on here -- maybe your processor isn't implementing an instruction that Asterisk or Dahdi is relying on. (It's my understanding that 64-bit processors don't fully implement the 32-bit instructions when in 64-bit mode, but I wouldn't swear to that.) Or maybe it's a library path problem -- something trying to use a 32-bit library instead of a 64-bit one, or vice versa. Try ldd on the binaries. ldd -v /usr/sbin/asterisk linux-vdso.so.1 = (0x7fff407ff000) libssl.so.1.0.0 = /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x7f72accc) libcrypto.so.1.0.0 = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x7f72ac911000) libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f72ac571000) libxml2.so.2 = /usr/lib/libxml2.so.2 (0x7f72ac216000) libdl.so.2 = /lib/x86_64-linux-gnu/libdl.so.2 (0x7f72ac012000) libpthread.so.0 = /lib/x86_64-linux-gnu/libpthread.so.0 (0x7f72abdf4000) libtinfo.so.5 = /lib/libtinfo.so.5 (0x7f72abbcd000) libm.so.6 = /lib/x86_64-linux-gnu/libm.so.6 (0x7f72ab949000) libresolv.so.2 = /lib/x86_64-linux-gnu/libresolv.so.2 (0x7f72ab72d000) libz.so.1 = /lib/x86_64-linux-gnu/libz.so.1 (0x7f72ab515000) /lib64/ld-linux-x86-64.so.2 (0x7f72acf1b000) Version information: /usr/sbin/asterisk: libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2 libresolv.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libresolv.so.2 libxml2.so.2 (LIBXML2_2.6.0) = /usr/lib/libxml2.so.2 libxml2.so.2 (LIBXML2_2.4.30) = /usr/lib/libxml2.so.2 libcrypto.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6 libpthread.so.0 (GLIBC_2.3.3) = /lib/x86_64-linux-gnu/libpthread.so.0 libpthread.so.0 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libpthread.so.0 libc.so.6 (GLIBC_2.8) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 libssl.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libssl.so.1.0.0 /lib/x86_64-linux-gnu/libssl.so.1.0.0: libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libcrypto.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 /lib/x86_64-linux-gnu/libcrypto.so.1.0.0: libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2 libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.7) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 /lib/x86_64-linux-gnu/libc.so.6: ld-linux-x86-64.so.2 (GLIBC_2.3) = /lib64/ld-linux-x86-64.so.2 ld-linux-x86-64.so.2 (GLIBC_PRIVATE) = /lib64/ld-linux-x86-64.so.2 /usr/lib/libxml2.so.2: libz.so.1 (ZLIB_1.2.2.3) = /lib/x86_64-linux-gnu/libz.so.1 libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2 libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6 libc.so.6 (GLIBC_2.7) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6
Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous
Hi virendra, Dialed same command.. I got below output ast18*CLI originate sip/test02 application dial == Using SIP RTP CoS mark 5 [Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid:192;tag=as417a5527' Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati virbh...@gmail.com wrote: Hi checked your debug like. Did you check that your SIP device ir registered with server ? if yes then dial below command from CLI *originate sip/test02 application dial* On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.comwrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com wrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Set Call Codec in extension.conf
Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
I loaded the latest 1.6 which gets slightly further and a core dump shows this, but its past my ability to interpret # gdb -se asterisk -c core | tee /tmp/backtrace.txt GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08 Copyright (C) 2011 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as x86_64-linux-gnu. For bug reporting instructions, please see: http://bugs.launchpad.net/gdb-linaro/... Reading symbols from /usr/sbin/asterisk...done. [New LWP 19322] [New LWP 19323] [New LWP 19324] [New LWP 19325] [New LWP 19326] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Core was generated by `asterisk -d -g -cvvv'. Program terminated with signal 4, Illegal instruction. #0 0x00500eab in tzload (name=optimized out, sp=0x1fc7950, doextend=1) at stdtime/localtime.c:424 424 static int tzload(const char *name, struct state * const sp, const int doextend) On 5/01/2012, at 12:13 AM, Duncan Turnbull wrote: On 4/01/2012, at 11:47 PM, A J Stiles wrote: For what it's worth, I once tried installing Asterisk on an old VIA C7 box; and it turns out that this processor, while detecting as an i686, doesn't implement the full i686 instruction set -- and Asterisk is trying to use one of the non-implemented instructions. Solution was to re-compile for i586. Thanks very much AJ That did appear as one of the few google comments I found but I couldn't figure out whether it applies The outputs are below if you can interpret them, I can see lm in the cpu proc info but don't know how to check for better compatibility It's just possible that something similar is going on here -- maybe your processor isn't implementing an instruction that Asterisk or Dahdi is relying on. (It's my understanding that 64-bit processors don't fully implement the 32-bit instructions when in 64-bit mode, but I wouldn't swear to that.) Or maybe it's a library path problem -- something trying to use a 32-bit library instead of a 64-bit one, or vice versa. Try ldd on the binaries. ldd -v /usr/sbin/asterisk linux-vdso.so.1 = (0x7fff407ff000) libssl.so.1.0.0 = /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x7f72accc) libcrypto.so.1.0.0 = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x7f72ac911000) libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f72ac571000) libxml2.so.2 = /usr/lib/libxml2.so.2 (0x7f72ac216000) libdl.so.2 = /lib/x86_64-linux-gnu/libdl.so.2 (0x7f72ac012000) libpthread.so.0 = /lib/x86_64-linux-gnu/libpthread.so.0 (0x7f72abdf4000) libtinfo.so.5 = /lib/libtinfo.so.5 (0x7f72abbcd000) libm.so.6 = /lib/x86_64-linux-gnu/libm.so.6 (0x7f72ab949000) libresolv.so.2 = /lib/x86_64-linux-gnu/libresolv.so.2 (0x7f72ab72d000) libz.so.1 = /lib/x86_64-linux-gnu/libz.so.1 (0x7f72ab515000) /lib64/ld-linux-x86-64.so.2 (0x7f72acf1b000) Version information: /usr/sbin/asterisk: libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2 libresolv.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libresolv.so.2 libxml2.so.2 (LIBXML2_2.6.0) = /usr/lib/libxml2.so.2 libxml2.so.2 (LIBXML2_2.4.30) = /usr/lib/libxml2.so.2 libcrypto.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6 libpthread.so.0 (GLIBC_2.3.3) = /lib/x86_64-linux-gnu/libpthread.so.0 libpthread.so.0 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libpthread.so.0 libc.so.6 (GLIBC_2.8) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 libssl.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libssl.so.1.0.0 /lib/x86_64-linux-gnu/libssl.so.1.0.0: libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6 libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6 libcrypto.so.1.0.0 (OPENSSL_1.0.0) = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 /lib/x86_64-linux-gnu/libcrypto.so.1.0.0: libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2 libc.so.6 (GLIBC_2.3) =
Re: [asterisk-users] Speech recognition in asterisk using google voice API
this looks great - is there any chance of coverting the googletts.agi to use flac as well ? Julian On 4 January 2012 09:06, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 07:51 AM, Bruce B wrote: And with recent version 14.3.2 I get: /usr/local/bin/sox FAIL formats: no handler for file extension `flac' -- speech-recog.agi: /usr/local/bin/sox failed: 512 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0 Regards, On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote: Very interesting. I just tried to get it to work but it complains about sox. Probably you used a different version of sox? *PBX-*CLI /usr/bin/sox: invalid option -- -* */usr/bin/sox: invalid option -- n* */usr/bin/sox: invalid option -- o* */usr/bin/sox: -r must be given a positive integer* * -- speech-recog.agi: /usr/bin/sox failed: 512* I am using: *Package sox-12.18.1-1.el5_5.1.i386 * Thanks, Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Speech recognition in asterisk using google voice API
On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote: this looks great - is there any chance of coverting the googletts.agi to use flac as well ? Julian In googletts.agi we get the voice data from google in mp3 and we convert it in a format that asterisk can read and playback (slin). If we store it in flac asterisk wont be able to read it natively and we would have to convert it each time we want to play it back to the user. In the speech recognition script we have to convert the voice data in flac before sending it to google because that's the accepted format. Is there some particular reason you want the googletts.agi data in flac? Lefteris Zafiris -- _ -- 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] Speech recognition in asterisk using google voice API
the only reason is that I didn't want to have to install sox. Lazy. that's all ;) Just another piece of software to find and install running on amazon ec2, is the best thing to download the source and compile sox ? Thanks Julian On 4 January 2012 14:18, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote: this looks great - is there any chance of coverting the googletts.agi to use flac as well ? Julian In googletts.agi we get the voice data from google in mp3 and we convert it in a format that asterisk can read and playback (slin). If we store it in flac asterisk wont be able to read it natively and we would have to convert it each time we want to play it back to the user. In the speech recognition script we have to convert the voice data in flac before sending it to google because that's the accepted format. Is there some particular reason you want the googletts.agi data in flac? Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Speech recognition in asterisk using google voice API
On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote: the only reason is that I didn't want to have to install sox. Lazy. that's all ;) Just another piece of software to find and install running on amazon ec2, is the best thing to download the source and compile sox ? Thanks It should be on your distro repos already. Lefteris Zafiris -- _ -- 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] Speech recognition in asterisk using google voice API
nope :( On 4 January 2012 14:29, Lefteris Zafiris zaf@gmail.com wrote: On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote: the only reason is that I didn't want to have to install sox. Lazy. that's all ;) Just another piece of software to find and install running on amazon ec2, is the best thing to download the source and compile sox ? Thanks It should be on your distro repos already. Lefteris Zafiris -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Which QSIG variant and profiles does asterisk support ?
Hello, Which QSIG (ECMA or ISO) variant and profiles does asterisk support ? (I could not find this info inside chan_dahdi.conf) Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rami
Hey, There is a new kid in town if you want to code in ruby. Use adhearsionhttps://github.com/adhearsion/adhearsion/wiki, it's a framework to make voice apps. On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV -- _ -- 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 -- Thanks Regards Gokulnath @8129845320 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. Please advise me to resolve this issue. -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem w/ PC port on Polycom 335
We did get this fixed. Turns out that my tech didn't reboot the phone after disabling the vlan configuration. He's new and still learning. Thank you for your time and suggestions. On Monday 02 January 2012 6:04:49 pm Jim DeVito wrote: Agreed. Check the switch for some kind of port security. Most of the time this would disable the interface if more than one MAC is present but you never know. Are there blinky lights on the pc? Also if provisioning via some sort of server check the MAC-boot log that the pgone uploads. Good Luck!! Thanks!! Jim. - Original message - Mike Diehl wrote: Usually, it just works... Any ideas? I've seen this before. One of our facilities have 'smart or managed' switches that have caused no ends of problems, including preventing computers plugged into the phones not having network access. You may want to review your switches. Doug http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Take care and have fun, Mike Diehl. -- _ -- 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] Set Call Codec in extension.conf
Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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] Set Call Codec in extension.conf
anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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] Set Call Codec in extension.conf
1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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] Set Call Codec in extension.conf
1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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 -- _ -- 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] Set Call Codec in extension.conf
I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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 -- _ -- 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] Set Call Codec in extension.conf
My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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] asterisk - AGI (perl) - sqlplus (oracle)
The module probably isn't readable/executeable from Asterisk From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, January 04, 2012 10:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. Please advise me to resolve this issue. -- Regards, Ahmed Munir Chohan -- _ -- 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] Set Call Codec in extension.conf
how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- 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 -- _ -- 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] Set Call Codec in extension.conf
Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?
Re: [asterisk-users] Set Call Codec in extension.conf
didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
Re: [asterisk-users] Set Call Codec in extension.conf
CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To:
Re: [asterisk-users] Set Call Codec in extension.conf
-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai l:fkha...@iconnecths.com) in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried
Re: [asterisk-users] Set Call Codec in extension.conf
You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai l:fkha...@iconnecths.com) in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before
Re: [asterisk-users] Set Call Codec in extension.conf
I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai l:fkha...@iconnecths.com) in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From:
Re: [asterisk-users] Set Call Codec in extension.conf
Any suggestion will be great From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... -- _ -- 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] Set Call Codec in extension.conf
Please post the sip.conf entries for 6000 and 6500. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai l:fkha...@iconnecths.com) in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
Re: [asterisk-users] Set Call Codec in extension.conf
there is nothing in sip.conf about what u asked but 6500 is a queue with following info [6500] fullname = testing strategy = rrmemory timeout = 15 wrapuptime = 15 autofill = no autopause = no joinempty = yes leavewhenempty = no reportholdtime = no maxlen = 0 musicclass = test member = SIP/6251 member = SIP/6252 member = SIP/6253 member = SIP/6254 now the user 6251 is a user with following info and caller 6000 [6000] username = 6000 transfer = yes mailbox = 6000 call-limit = 100 type = peer fullname = 6000 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6000 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 callcounter = yes [6251] username = 6251 transfer = yes mailbox = 6251 call-limit = 100 type = peer fullname = 6251 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6251 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = yes hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Please post the sip.conf entries for 6000 and 6500. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai l:fkha...@iconnecths.com) in
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On Wednesday 04 January 2012, Duncan Turnbull wrote: I loaded the latest 1.6 which gets slightly further and a core dump shows this, but its past my ability to interpret # gdb -se asterisk -c core | tee /tmp/backtrace.txt GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08 Copyright (C) 2011 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as x86_64-linux-gnu. For bug reporting instructions, please see: http://bugs.launchpad.net/gdb-linaro/... Reading symbols from /usr/sbin/asterisk...done. [New LWP 19322] [New LWP 19323] [New LWP 19324] [New LWP 19325] [New LWP 19326] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Core was generated by `asterisk -d -g -cvvv'. Program terminated with signal 4, Illegal instruction. #0 0x00500eab in tzload (name=optimized out, sp=0x1fc7950, doextend=1) at stdtime/localtime.c:424 424static int tzload(const char *name, struct state * const sp, const int doextend) It's a bit beyond my depth too, but I'd start with a look at localtime.c in the Asterisk source tree. It might simply be trying to include something that isn't present on your system. If you stick a /* harmless comment */ in this file and re-save it, this will give the file a new modification time. Then run `make` again. It will recompile just localtime.c (this being the only source file that has changed since the last time make was run) -- now watch very closely for errors. Answers come *after* questions. -- _ -- 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] Rami
Is this freeware, or a module which you can include in your ruby code? Or is it a complete framework? On 04 Jan 2012, at 5:31 PM, gokulnath wrote: Hey, There is a new kid in town if you want to code in ruby. Use adhearsionhttps://github.com/adhearsion/adhearsion/wiki, it's a framework to make voice apps. On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote: Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://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 -- Thanks Regards Gokulnath @8129845320 -- _ -- 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] Set Call Codec in extension.conf
What about the allow/disallow lines in sip.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf there is nothing in sip.conf about what u asked but 6500 is a queue with following info [6500] fullname = testing strategy = rrmemory timeout = 15 wrapuptime = 15 autofill = no autopause = no joinempty = yes leavewhenempty = no reportholdtime = no maxlen = 0 musicclass = test member = SIP/6251 member = SIP/6252 member = SIP/6253 member = SIP/6254 now the user 6251 is a user with following info and caller 6000 [6000] username = 6000 transfer = yes mailbox = 6000 call-limit = 100 type = peer fullname = 6000 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6000 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 callcounter = yes [6251] username = 6251 transfer = yes mailbox = 6251 call-limit = 100 type = peer fullname = 6251 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6251 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = yes hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Please post the sip.conf entries for 6000 and 6500. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm ) in new stack -- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB OUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT BOUND=gsm) in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format
Re: [asterisk-users] Set Call Codec in extension.conf
allow=all From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf What about the allow/disallow lines in sip.conf? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
Hi, I installed the modules in asterisk user home directory with read and excitable permissions for asterisk but still my AGI not working. Please provide me other advise to resolve this issue. Date: Wed, 4 Jan 2012 11:30:34 -0600 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com Content-Type: text/plain; charset=us-ascii The module probably isn't readable/executeable from Asterisk From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, January 04, 2012 10:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. Please advise me to resolve this issue. -- Regards, Ahmed Munir Chohan - Regards, Ahmed Munir Chohan -- _ -- 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] Set Call Codec in extension.conf
Both sides? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf allow=all From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf What about the allow/disallow lines in sip.conf? -- _ -- 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] Set Call Codec in extension.conf
yup and video support is yes From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Both sides? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf allow=all From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf What about the allow/disallow lines in sip.conf? -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
What are the permissions on the module you are trying to run? (ls -l /var/lib/asterisk/agi-bin/module) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, January 04, 2012 12:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) Hi, I installed the modules in asterisk user home directory with read and excitable permissions for asterisk but still my AGI not working. Please provide me other advise to resolve this issue. Date: Wed, 4 Jan 2012 11:30:34 -0600 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com Content-Type: text/plain; charset=us-ascii The module probably isn't readable/executeable from Asterisk From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, January 04, 2012 10:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle) Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. Please advise me to resolve this issue. -- Regards, Ahmed Munir Chohan - Regards, Ahmed Munir Chohan -- _ -- 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] Speech recognition in asterisk using google voice API
Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Best, -- _ -- 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] Speech recognition in asterisk using google voice API
Does anyone know what languages are supported? -Original Message- From: Bruce B bruceb...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 4 Jan 2012 13:25:18 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Speech recognition in asterisk using google voice API -- _ -- 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] Speech recognition in asterisk using google voice API
Wow - nice! A few quick questions: 1. How long can the recording be for translation? 2. Any limitation on how much text the return (transcribed) variable can hold? 3. Any commercial / terms of use limitations? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B [bruceb...@gmail.com] Sent: Wednesday, January 04, 2012 1:25 PM To: Asterisk Users List Subject: Re: [asterisk-users] Speech recognition in asterisk using google voice API Note to self: Never release anything asterisk related without testing on RHEL/Centos 5 Thank you for reporting this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Best, -- _ -- 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] Rami
On Wed, 4 Jan 2012, Arjan Kroon | Mobillion wrote: Is this freeware, or a module which you can include in your ruby code?Or is it a complete framework? Is this list faster than Google? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On Wed, 4 Jan 2012, A J Stiles wrote: If you stick a /* harmless comment */ in this file and re-save it, this will give the file a new modification time. Then run `make` again. It will recompile just localtime.c (this being the only source file that has changed since the last time make was run) -- now watch very closely for errors. The 'touch' command will update the file's access and modification times* without the risk of trashing something in the file. *) Command line parameters can select just the access or the modification time. The default is both. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Speech recognition in asterisk using google voice API
On Wed, Jan 4, 2012 at 8:47 PM, Michelle Dupuis mdup...@ocg.ca wrote: Wow - nice! A few quick questions: 1. How long can the recording be for translation? At the moment the recording timeout is set at 15sec. I haven't tested yet the max length of voice data ta google accepts (all this voice recognition stuff is undocumented). I have read that it is between 10-20 seconds but havent really went to test this yet. On my todo list is to add the option to cut the sound data in smaller chunks before sending them to google and get rid of the recording length limitations. 2. Any limitation on how much text the return (transcribed) variable can hold? This better be answered by the astsrisk devs but empirically talking i have loaded in dialplan variables really big chunks of text (like the complete gpl license) without having any problems. 3. Any commercial / terms of use limitations? This is a gray area at the moment. Voice recognition is undocumented in google's API and i guess not officially supported yet. I hope it gets covered by the general TOS of google services: http://www.google.com/accounts/TOS Lefteris Zafiris -- _ -- 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] Speech recognition in asterisk using google voice API
On Wed, Jan 4, 2012 at 8:27 PM, isr...@gmail.com wrote: Does anyone know what languages are supported? For sure english and spanish, since its undocumented i don't have a complete list yet. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
Un-top-posting... On Wed, 4 Jan 2012, Ahmed Munir wrote: I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. It usually boils down to PATHs, environment variables, permissions, missing script interpreters, etc. When you execute your AGI from the command line, are you passing a valid AGI environment to the AGI via STDIN? If not, you may be violating the AGI protocol, which, is probably not why your 'AGI' is not executing, but will bite you some time in the future. When you say 'it don't get executed' do you mean that Asterisk cannot locate the AGI at all or do you mean it does not execute as you expect? On Wed, 4 Jan 2012, Ahmed Munir wrote: I installed the modules in asterisk user home directory with read and excitable permissions for asterisk but still my AGI not working. Asterisk looks for AGI executables in the 'astagidir' (AKA ASTAGIDIR) directory which is usually /var/lib/asterisk/agi-bin/. This can be set in asterisk.conf. The path to this file can be specified on the command line used to start asterisk. It would be unusual for ASTAGIDIR to be set the asterisk user's home directory. It would also be unusual for the asterisk user's home directory to be set to ASTAGIDIR. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Speech recognition in asterisk using google voice API
Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Google accepts sound files at any sampling rate (up to 44.1kHz) so if you can use some wideband codec ( eg g722) It can greatly improve the sound quality and the detection rates. For now the script supports 8kHz and 16kHz sampling rates for recording and it can be set by editing the scripts user defined parameters ( the variable $samplerate). Anything that improves the recording sound clarity will help, a good phone, low background noise level etc. I have also read that normalizing the recording and setting the gain to -5 db improves detection rates. I m experimenting with this at the moment and there will be some new code soon (as soon as i get sox working in RHEL/Centos 5 :P ). Lefteris Zafiris -- _ -- 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] From address missing 'sip:', using it anyway
Hello, I see the following error in the logs [Jan 4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address missing 'sip:', using it anyway Does anybody know how to stop this error? It does not seem to be affecting performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I have google it but empty! Thanks, Celso -- _ -- 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] Speech recognition in asterisk using google voice API
On 1/4/2012 2:26 PM, Lefteris Zafiris wrote: Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Google accepts sound files at any sampling rate (up to 44.1kHz) so if you can use some wideband codec ( eg g722) It can greatly improve the sound quality and the detection rates. For now the script supports 8kHz and 16kHz sampling rates for recording and it can be set by editing the scripts user defined parameters ( the variable $samplerate). Anything that improves the recording sound clarity will help, a good phone, low background noise level etc. I have also read that normalizing the recording and setting the gain to -5 db improves detection rates. I m experimenting with this at the moment and there will be some new code soon (as soon as i get sox working in RHEL/Centos 5 :P ). This is really spectacular. Thanks. I'm running Fedora 15, so I can use flac or sox. Any reason to prefer one over the other? 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] Speech recognition in asterisk using google voice API
wow i just tried in hebrew and i'll say just 1 word WOW On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote: On 1/4/2012 2:26 PM, Lefteris Zafiris wrote: Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, hello. I think there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. Probably Google sees the pattern?! What are some of the other tricks (if any) or consideration that one should make while creating a strong speech recognition enabled IVR? Google accepts sound files at any sampling rate (up to 44.1kHz) so if you can use some wideband codec ( eg g722) It can greatly improve the sound quality and the detection rates. For now the script supports 8kHz and 16kHz sampling rates for recording and it can be set by editing the scripts user defined parameters ( the variable $samplerate). Anything that improves the recording sound clarity will help, a good phone, low background noise level etc. I have also read that normalizing the recording and setting the gain to -5 db improves detection rates. I m experimenting with this at the moment and there will be some new code soon (as soon as i get sox working in RHEL/Centos 5 :P ). This is really spectacular. Thanks. I'm running Fedora 15, so I can use flac or sox. Any reason to prefer one over the other? 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-**usershttp://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] Failed to authenticate on INVITE to Anonymous
On 1/4/2012 4:37 AM, Jayesh Labade wrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com wrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.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 Try: register = test02:test02@192.168.1.55/s 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] Speech recognition in asterisk using google voice API
On Wed, 04 Jan 2012 14:48:22 -0500 sean darcy seandar...@gmail.com wrote: This is really spectacular. Thanks. I'm running Fedora 15, so I can use flac or sox. Any reason to prefer one over the other? sean We have to convert the voice data to flac format before sending them to google, this can be done by both sox and flac encoder. For now the script uses flac encoder for compatibility with older distros (mainly RHEL 5). Sox is a bit more flexible and also gives you the option to edit the sound data (normalizing, changing levels etc). Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On 5/01/2012, at 8:06 AM, Steve Edwards wrote: On Wed, 4 Jan 2012, A J Stiles wrote: If you stick a /* harmless comment */ in this file and re-save it, this will give the file a new modification time. Then run `make` again. It will recompile just localtime.c (this being the only source file that has changed since the last time make was run) -- now watch very closely for errors. The 'touch' command will update the file's access and modification times* without the risk of trashing something in the file. *) Command line parameters can select just the access or the modification time. The default is both. Touch seemed safer but I didn't see any errors :/usr/src/asterisk-1.6.2.22# make [CC] stdtime/localtime.c - stdtime/localtime.o [LD] abstract_jb.o acl.o adsistub.o aescrypt.o aeskey.o aestab.o alaw.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autoservice.o bridging.o callerid.o cdr.o channel.o chanvars.o cli.o config.o cryptostub.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o logger.o manager.o md5.o netsock.o pbx.o plc.o poll.o privacy.o rtp.o say.o sched.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk +- Asterisk Build Complete -+ + Asterisk has successfully been built, and + + can be installed by running: + + + +make install + +---+ I haven't found anything obvious in the debug stuff although I am not familiar enough to be sure Thanks very much Unless there is something obvious I am thinking I will revert to either an earlier OS or maybe 32 bit - although that seems excessive Cheers Duncan -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] question sangoma vs digium
Hi! Hello! I wanted to know if you have experienced problems installing both a Sangoma and a Digium card in the same Server. Thnks a lot! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
The Asterisk Development Team is pleased to announce the first release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0. 2.6.0 is a feature release which: - Adds support for the TE820 8-span card to the wct4xxp driver. - Decrease load time of analog cards supported by the wctdm24xxp driver. - Adds sysfs object model to facilitate persistent span numbering and early loading of modules (NOTE: by default this release still behaves like previous releases with regards to span numbering assignment). - dahdi_pcap tool is now included in DAHDI-tools but not compiled by default since it depends on a currently unsupported interface in DAHDI-Linux. It is intended that in future releases this will be compiled by default. Issues closed in this release: DAHTOOL-49: adding pcap support to Dahdi (Reported by: Torrey Searle) DAHLIN-258: weird sound with a native bridged isdn-bri connection (Reported by: Daniel) DAHLIN-264: xpp: E1 CAS multiframe bits not properly set DAHDI-Linux 2.6.0, DAHDI-Tools 2.6.0, and DAHDI-Linux-Complete 2.6.0+2.6.0 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete The DAHDI-Linux shortlog of changes that are not in 2.5.0.2: Doug Bailey: wctdm24xxp, wcte12xp: Update VPMOCT032 firmware to 1.12.0. Tzafrir Cohen: Avoid building PCI devices if kernel has no PCI xpp: Allow up to 128 Astribanks on a system xpp: increase command queue length to 1500 xpp: USB_FW rev 10085: fix regression from r10013 xpp: PIC_TYPE_1 rev 9841: followup to r10013 bugfix: off-by-one in span assignment xpp: USB firmware r9964: minor bugfixes xpp: bugfix: clear NOTOPEN span alarm on assign xpp: bugfix -- manage xpd refcount for EC module xpp: Adaptations for E-Main-3 xpp: remove leftovers of old XPD_STATE method README: Minor additions regarding pinned-spans README: initial update for span assignments dahdi: Add error messages in dahdi_ioctl_chanconfig. xpp: fix FXS D DTMF detection (not zero) xpp: fix bashism in xpp_debug live_dahdi: optionally generate FreePBX DB entries Matthew Fredrickson: wct4xxp: Add support for TE820 and VPMOCT256. Russ Meyerriecks: wct4xxp: Remove vpm400 support. wct4xxp: Revise vpm struct due to product name changes wct4xxp: Handle incorrect vpm module/card pairings wct4xxp: minor: Removed unnecessary instrumentation wct4xxp: Expose serial number in dahdi_device and kernel log. wct4xxp: Add field upgradable firmware support for TE820. wcte12xp, wctdm24xxp: Remove frowny face from vpmoct032 error message Oron Peled: xpp: BRI: batch D-Channel packets to fix frag. xpp: BRI: split multibyte functionality xpp: BRI: remove trivial BRISTUFF wrappers xpp: BRI: remove legacy BRISTUFF code xpp: bad module_put() when too many Astribanks DAHDI-linux: Fix surprise removal problems xpp: BRI: fix timing priority calculation xpp: FXS: mwi and search_fsk fixes xpp: PRI: restore pri_protocol to R/W: xpp: pri: fix RS1 init in E1 CAS mode xpp: fxs: demote SETPOLARITY message to DBG() xpp: silence some bad ioctl() reporting xpp: restore backward compat dahdi_registration Extra debugging aids and messages xpp: cleanup some printk()'s added 'basechan' and 'channels' attributes to spans dahdi: Give userspace a chance to respond to surprise removal. xpp: Remove dahdi_autoreg parameter: xpp: more informative span description: xpp: make unregistration safer (idempotent) xpp: adapt to 'location' attribute removal: xpp: PRI: use DAHDI new set_spantype() method dahdi: Expose spans in sysfs. dahdi: dahdi_is_analog_span() - dahdi_is_digital_span() dahdi: start handling surprise device removal. Shaun Ruffell: wctdm24xxp: Fix bug if hook state on FXS changes before channel configuration. wct4xxp: Reduce time spent waiting for auth done bit on TE820. wct4xxp: Fail startup if not generating interrupts. dahdi: Return dahdi_span_ops.startup callback errors to userspace. wctdm24xxp: Do not call voicebus_release() before wctdm_back_out_gracefully() dahdi: #include linux/module.h in dahdi/kernel.h and GpakCust.h wctc4xxp: Replace 'ndo_set_multicast_list' with 'set_rx_mode' wctdm24xxp: Wait for background threads to complete on failed load. dahdi: Unregister dahdi_device from sysfs if we fail to auto assign spans. dahdi: Fix typo in previous commit which forced some spans to always fail assignment. dahdi:
Re: [asterisk-users] asterisk 1.8 codec negotiation
On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message multiple audio streams not supported in the log. multiple audio streams != multiple audio codecs. For some reason Asterisk is receiving an INVITE with an offer for more than one audio stream (m=audio), and that is not supported. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] NAT/IPTABLES workarounds
On 01/03/2012 10:03 AM, Patrick Lists wrote: On 03-01-12 16:24, Danny Nicholas wrote: Hello List, I work in an environment where I have to request IPTABLES changes rather than doing them myself. Is there a way to utilize my SSH (port 22) to get a functional (and with good sound) Asterisk installation with multiple channels up without requesting the 5060(SIP) 5061 (TLS) and UDP/RTP (usually 10001-2) IPTABLES allowances? Not with SIP as it needs a port for signaling (usually 5060) and RTP ports for sending the actual voice packets. So for SIP you will always need multiple ports. If you can use IAX then you could use port 22 as IAX only needs one port. The question is how are you going to SSH into the box if you use the SSH port for Asterisk? It is not practical (although not impossible) to run UDP over an SSH tunnel. Since VoIP media is generally transported over UDP, this will be a major obstacle. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Anyone have a reliable T.38 Solution
On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk won't start - trap invalid opcode
DT == Duncan Turnbull dun...@e-simple.co.nz writes: DT I have a new install of asterisk 1.8.8.1 on ubuntu server DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux DT The only errors I can see are limited - I also stopped wan router and dahdi and I still get DT ~# asterisk -cvv DT Illegal instruction What does /proc/cpuinfo say? (Just the first chunk is enough.) Try running asterisk is gdb: :; gdb asterisk (gdb) run -cvvddd When it dies, try: (gdb) bt full (gdb) disasemble /m You may also want to recompile asterisk after turing on: DONT_OPTIMIZE DEBUG_THREADS BETTER_BACKTRACES in the Compiler Flags section of make menuselect. The gdb output if you do that may be more comprehensible. Either way run gdb from the asterisk src directory. When you find the point where it crashed, you can discover what the illegal instruction is. I suspect your compile may expect a more recent cpu than you have, and may use sse instructions which it doesn't support. A disassembly around the failing instruction will confirm whether that is true and which instruction it is. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] Speech recognition in asterisk using google voice API
Fresh code is out! The use of sox can be now optionally enabled by the user if the system has a recent version of the program (won't work in RHEL/Centos 5) This is done by editing the script and setting the variable 'use_sox'. When sox is used the audio gets normalized, low frequency noise (100Hz) is removed and also possible DC offset is corrected. Those are supposed to improve the recognition results(?). The settings are still a bit experimental, feel free to play with them and report what settings improved your results. get the new version here: https://github.com/downloads/zaf/asterisk-speech-recog/asterisk-speech-recog-0.3.tar.gz Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
On 5/01/2012, at 12:21 PM, James Cloos wrote: DT == Duncan Turnbull dun...@e-simple.co.nz writes: DT I have a new install of asterisk 1.8.8.1 on ubuntu server DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux DT The only errors I can see are limited - I also stopped wan router and dahdi and I still get DT ~# asterisk -cvv DT Illegal instruction What does /proc/cpuinfo say? (Just the first chunk is enough.) Try running asterisk is gdb: :; gdb asterisk (gdb) run -cvvddd When it dies, try: (gdb) bt full (gdb) disasemble /m You may also want to recompile asterisk after turing on: DONT_OPTIMIZE DEBUG_THREADS Hi James I think the DONT_OPTIMIZE flag made a difference, the system is not crashing anymore I am going to test it, and see if its really back, the other detail looked fairly similar to the core dump output in previous emails but there wasn't anything I could easily discern I will let you all know how it turns out - thanks everyone Cheers Duncan BETTER_BACKTRACES in the Compiler Flags section of make menuselect. The gdb output if you do that may be more comprehensible. Either way run gdb from the asterisk src directory. When you find the point where it crashed, you can discover what the illegal instruction is. I suspect your compile may expect a more recent cpu than you have, and may use sse instructions which it doesn't support. A disassembly around the failing instruction will confirm whether that is true and which instruction it is. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 codec negotiation
Kevin P. Fleming kpflem...@digium.com wrote: On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message multiple audio streams not supported in the log. multiple audio streams != multiple audio codecs. For some reason Asterisk is receiving an INVITE with an offer for more than one audio stream (m=audio), and that is not supported. OK, but if I have a phone or in my case a server which offers a choice of codecs, why can't asterisk just pick the ones it has rather than reject the call? Is there a way to do this correctly as far as asterisk is concerned? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] question sangoma vs digium
hello: i think it can be done, please refer this link: http://wiki.sangoma.com/Asterisk-FAQ#Digium Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Wed, 4 Jan 2012 18:47:28 -0200 From: agustina.berre...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] question sangoma vs digium Hi! Hello! I wanted to know if you have experienced problems installing both a Sangoma and a Digium card in the same Server. Thnks a lot! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] 回覆︰ dialplan - dial command - custom ringtone
my config: hardphone - pstn gateway - asterisk - pstn gateway - hardphone i am using asterisk 1.4.xx w option is Dial is for recording. how does it related to ringtone? pls advise. 從︰ Carlos Rojas crt.ro...@gmail.com 收件人︰ Qqblog Qqblog qqb...@ymail.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2012年01月3日 (週二) 8:42 PM 主題︰ Re: [asterisk-users] dialplan - dial command - custom ringtone Hello Do you use hard phone or softphone? In many ip phones you can change the ring tones or use w option in Dial command Regards On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote: i could add r option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -- _ -- 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] Anyone have a reliable T.38 Solution
On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or 1.8? Glad to know that the PAP2T has a solid T.38 implementation! -Matt -- _ -- 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] Anyone have a reliable T.38 Solution
I'm using 1.8, but also have 1.4 and 1.2 installs using the same configuration. Regards David. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Thursday, 5 January 2012 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anyone have a reliable T.38 Solution On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or 1.8? Glad to know that the PAP2T has a solid T.38 implementation! -Matt -- _ -- 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] Server-to-server BLF
Hi Kevin, Thanks for your suggestion. On the website of OpenAIS, it seems that it is not supported anymore and their download links (FTP and SVN) are broken (been trying it for about a month now). Is it still possible to use OpenAIS method? The other solution on the wiki is using XMPP which is for jabber. IMHO, it means that the XMPP solution can't be used on SIP peers, right? Regards, Ronald On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 11/16/2011 04:18 AM, Ronald Cepres wrote: Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Here is one way: https://wiki.asterisk.org/**wiki/display/AST/Distributed+** Device+State+with+AIShttps://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS There are other methods documented on the wiki as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://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] Set Call type in dial plan
Hi, Sorry for late reply. Hope you've already found out something about it. What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk. See these pages: http://www.voip-info.org/wiki/view/Asterisk+variables https://issues.asterisk.org/view.php?id=13243 Regards, Sammy On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib fkha...@iconnecths.com wrote: thats excatly what I want, can u plz give me the command, I want to choose only ulow From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [ govoi...@gmail.com] Sent: Tuesday, January 03, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan Hi, For such call you just need to select the outbound codec before the dial() app. choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib fkha...@iconnecths.com mailto:fkha...@iconnecths.com wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: sip:6097@192.168.21.102mailto:sip%3A6097@192.168.21.102 ;tag=1857098215 To: sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 Contact: sip:6097@192.168.21.193:52933 ;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: sip:192.168.21.102:5060;lr;transport=udp Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- 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 -- _ -- 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] DIALSTATUS Values
Can anybody please reply on this? Regards, Kamlesh From: kamlesh_...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 27 Dec 2011 09:49:21 + Subject: Re: [asterisk-users] DIALSTATUS Values Hello, After investing some time, I could come to know the reason for not getting the data value is that if I use system command with any of asterisk cli command as given below, data value is returned blank. $output=system(/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d / | grep '100' ) Could you please suggest now how to rectify this? Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 12:27:19 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- _ -- 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] Using Asterisk as a softphone
Hi, one reason for having that robotic voice could be improper codecs others include low CPU processing power, memory not free etc. I once had the same kind of issue with VAD(voice activity detection) turned ON from my service providers equipment so my asterisk was performing poorly with VAD. Asterisk version and its codec play more important role. Regards, Sammy On Tue, Jan 3, 2012 at 6:34 PM, Christian Jaeger chr...@gmail.com wrote: Hello I'm using softphones as my only 'landline' phone service for almost 3 years now (Diamondcard and now voip.ms), so far using SIP (and mostly Twinkle). Also, I'm using Linux (Debian) as my choice of desktop OS. Also, sometimes I'm in networks with badly behaving NAT routers (for some time I used openvpn to solve this unreliably, then I ended up using 3G instead of wifi while in Canada, but now I'm abroad and don't have 3G). I'm now sufficiently fed up with SIP to give IAX2 another try. I want a softphone solution that: * works on Linux (Debian) * works reliably (e.g. remain connected for incoming calls, work with shitty NAT routers) * preferably encrypts both signalling and voice (dunno if voip.ms supports it, I might use a proxy asterisk instance on an own server instead) * properly handles audio with the 8000 samples/second dictated by the POTS systems (ALSA combined with some hardware (like both of my laptops) doesn't do proper lowpass filtering for mic input, so I will have to either use OSS or PulseAudio or rely on Asterisk doing proper downsampling in software). Asterisk seems to fit the first three; I'll happily build a GUI on top if this turns out to be a stable solution. My problems right now: - when I issue console dial without a number, it plays a recording with a woman's voice, and I can understand what is being said, but it sounds very garbled, like modulated with some about 20 Hz signal (a bit like a robot voice). What could be the problem? (Not using pulseaudio; +- default configuration.) One hypothesis I have is that it uses a too small buffer somewhere. - I don't understand how the extensions stuff is working. voip.ms wiki told me to create sections named [voipms], but how do I switch to 'default'? tie*CLI console dial 4443 No such extension '4443' in context 'default' tie*CLI console dial 04443 No such extension '04443' in context 'default' tie*CLI console dial 004443 No such extension '004443' in context 'default' - I haven't found anyone in google who tried to do the same as me, except http://www.junghanns.net/en/asteriskassoftphone.html but that doesn't lead me far (and the patch linked is unavailabe). Has anyone here done what I envision, or seen some docs specifically matching my use case? Thanks Christian. -- _ -- 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] From address missing 'sip:', using it anyway
Hi, The server or client application that is sending you sip packets is missing the sip: string in from header. You should have it sorted out because if that header goes to some external equipment the call may fail because of this. Regards, Sammy On Thu, Jan 5, 2012 at 12:44 AM, motty.cruz motty.c...@gmail.com wrote: Hello, I see the following error in the logs [Jan 4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address missing 'sip:', using it anyway Does anybody know how to stop this error? It does not seem to be affecting performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I have google it but empty! Thanks, Celso -- _ -- 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] DIALSTATUS Values
This works fine for me, $dialstatus = $agi-get_variable(DIALSTATUS); $cdr['dialstatus'] = $dialstatus['data']; Try as it is, I believe it's because of concatenation. Regards, Zohair Raza On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield t...@softins.co.uk wrote: In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- _ -- 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] NAT/IPTABLES workarounds
Are you talking about having an SSH tunnel and route your SIP traffic through it !!? On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/03/2012 10:03 AM, Patrick Lists wrote: On 03-01-12 16:24, Danny Nicholas wrote: Hello List, I work in an environment where I have to request IPTABLES changes rather than doing them myself. Is there a way to utilize my SSH (port 22) to get a functional (and with good sound) Asterisk installation with multiple channels up without requesting the 5060(SIP) 5061 (TLS) and UDP/RTP (usually 10001-2) IPTABLES allowances? Not with SIP as it needs a port for signaling (usually 5060) and RTP ports for sending the actual voice packets. So for SIP you will always need multiple ports. If you can use IAX then you could use port 22 as IAX only needs one port. The question is how are you going to SSH into the box if you use the SSH port for Asterisk? It is not practical (although not impossible) to run UDP over an SSH tunnel. Since VoIP media is generally transported over UDP, this will be a major obstacle. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://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] Where are the fax instructions?
Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten = 306,1,NoOp(Fax transmission) same = n,Set(FAXOPT(gateway)=yes) same = n,Dial(DAHDI/3)-FXS port to fax machine same = n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -- Asterisk with TDM410 -- FXS -- Analog fax machine I am totally lost about the use of this new gateway module in the dialplan. I think it loads ok: CLI fax show capabilities Registered FAX Technology Modules: Type: Spandsp Description : Spandsp FAX Driver Capabilities: SEND RECEIVE T.38 G.711 1 registered modules Also I have the FFA manual, which I couldn't understand. I think FAXOPT is common to both, but still not sure how to put them together. Where can I find documentation about configuring the call flow described? Or some insight will also be appreciated. Here is my sip peer config: [105](headquarters) ;zoiper phone type=friend secret= mailbox=105@default t38pt_udptl = yes Dahdi: ;FXS Modules group = 2 signalling = fxo_ks context = interno channel = 3-4 faxdetect = both Finally, a verbose output: == Using SIP RTP CoS mark 5 -- Executing [606@intern:1] NoOp(SIP/105-0002, Fax Transmission) in new stack -- Executing [606@intern:2] Set(SIP/105-0002, FAXOPT(gateway)=yes) in new stack [Jan 5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write: channel 'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled! -- Executing [606@intern:3] Dial(SIP/605-0002, DAHDI/3) in new stack -- Called DAHDI/3 -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 answered SIP/605-0002 -- Hanging up on 'DAHDI/3-1' -- Hungup 'DAHDI/3-1' == Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-0002' Thanks in advance for any help *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
I guess this is a permissions issue. Make sure your agi script has execute permissions (755) and it belongs to asterisk:asterisk . for that you need: chmod 755 /var/lib/asterisk/agi-bin/agi-script-name.agi chown asterisk:asterisk /var/lib/asterisk/agi-bin/agi-script-name.agi Regards, LL On 1/4/2012 7:19 PM, Steve Edwards wrote: Un-top-posting... On Wed, 4 Jan 2012, Ahmed Munir wrote: I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get executed. It usually boils down to PATHs, environment variables, permissions, missing script interpreters, etc. When you execute your AGI from the command line, are you passing a valid AGI environment to the AGI via STDIN? If not, you may be violating the AGI protocol, which, is probably not why your 'AGI' is not executing, but will bite you some time in the future. When you say 'it don't get executed' do you mean that Asterisk cannot locate the AGI at all or do you mean it does not execute as you expect? On Wed, 4 Jan 2012, Ahmed Munir wrote: I installed the modules in asterisk user home directory with read and excitable permissions for asterisk but still my AGI not working. Asterisk looks for AGI executables in the 'astagidir' (AKA ASTAGIDIR) directory which is usually /var/lib/asterisk/agi-bin/. This can be set in asterisk.conf. The path to this file can be specified on the command line used to start asterisk. It would be unusual for ASTAGIDIR to be set the asterisk user's home directory. It would also be unusual for the asterisk user's home directory to be set to ASTAGIDIR. -- _ -- 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