Re: [asterisk-users] asterisk-users Digest, Vol 59, Issue 28
Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS (was asterisk-users Digest, Vol 59, Issue 28)
On Fri, Jun 12, 2009 at 8:51 AM, Kengie Hokengiepa...@gmail.com wrote: I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. If the ATA is connected to a router you can use DynDNS if the roiuter supports it; This will update a domain name of your choice with the dynamic IP address. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI connection with ZTE exchange over T1 PRI
Hi My production Asterisk has these frequent errors when connected to the ZTE type exchange... Jun 9 10:48:24 NOTICE[1984] chan_zap.c: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 followed by.. Jun 9 11:46:42 WARNING[1984] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! After a few moments later, the D-channels comes up automatically. My settings are as follows:- *zaptel.conf* span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us *zapata.conf* [trunkgroups] [channels] language=en context=pstn_default switchtype=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=400 rxgain=0.5 txgain=0.5 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived channel = 1-23 Any suggestions? Thanks, Si ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Aastra phones provisioning
2009/6/11 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
Aastra SIP phones can be auto-provisioned when config files are stored in
a
specific TFTP subdirectory instead of TFTP root directory.
For instance, TFTP root directory is /srv/tftp.
When config files are stored in /srv/tftp, a new Aastra can find its
config
files.
When config files are stored in /srv/tftp/aastra, a new Aastra can't find
its config files.
I tried to using DHCP root-path option to tell Aastra phones to search
the
right subdirectory, but it doesn't seem to work.
https://svn.amooma.com/gemeinschaft/trunk/usr/share/doc/gemeinschaft/misc/dhcpd-3-example.conf:
class Aastra {
match if substring(hardware, 1, 3) = 00:08:5D;
option tftp-server-name
http://192.168.1.130/gemeinschaft/prov/aastra/;;
# Aastra does not support any :port in the URL, not even :80
# for firmware app versions 2.1.2
}
That should work analogously for TFTP.
Thanks : I'll give it a try next monday and report my findings here.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Queue Problem
I am running Asterisk 1.4.21.2 For reception, I defined a simple queue with one SIP phone as the only member. When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it is 0. If it is 0, then I will playback a message to tell the caller to be patient and then do a Queue(queue-name). If QUEUE_WAITING_COUNT is zero, then I will just Queue(queue-name, r) to ring the receptionist phone without playing any message. A problem arises if the receptionist is talking to someone on the phone. In this scenario, QUEUE_WAITING_COUNT is also zero but I will need to playback a pls-be-patient message as well. So, I need to find out whether the receptionist phone is busy even if QUEUE_WAITING_COUNT = 0. Do you know if there is anyway, without dialling a SIP channel, I can check if a SIP extension is engaged or not? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple PRI's in one group ..how??
Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. Now i have zaptel.conf file created as follows -zaptel.conf-- # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 span=2,1,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Global data loadzone= us defaultzone = us --- As i see from some web examples i made my zapata.conf as follows.. ---zapata.conf [channels] language=en context=default switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 - First of all, im not sure if that config is correct or not... Any corrections are welcome about creating group. Second.. What should i write to my freepbx zap configurator. g1 as zap identifier??? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday 12th June @ 12 Noon EDT: VoIP Users Conference Skype to ZipDX
Hi all, In about 4 hours from this writing, the G.722 conference bridge will be brought up, the Talkshoe G.711 also, so you can call in via SIP, PSTN or Skype (experimental) http://vuc.me for all the gritty details IRC #voip-users-conference anytime today We'll also be talking about hosted PBX systems and what they might need to bring telephone into not the 20th, but the 21st century. It seems like only yesterday, but it was 130 years ago in 1879, that the Bell company acquired Edison's patents for the carbon microphone from Western Union. This made the telephone practical for long distances and it was no longer necessary to shout to be heard at the receiving telephone. Funny, but on most cell phones it is again necessary to shout to be heard; See you in a few at sip:7463#2262...@proxy.ideasip.com or sip:200...@login.zipdx.com or skype:pfh-zdx?calltopic=200901 /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple PRI's in one group ..how??
On Fri, Jun 12, 2009 at 11:49:02AM +0300, Oguzhan Kayhan wrote: Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. What do you mean by group? (I can think of two different answers depending on what you mean. either group= in chan_dahdi.conf or [trunkgroups] in the same file. Both are completely different) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple PRI's in one group ..how??
On Fri, Jun 12, 2009 at 8:13 AM, Tzafrir Cohentzafrir.co...@xorcom.com wrote: On Fri, Jun 12, 2009 at 11:49:02AM +0300, Oguzhan Kayhan wrote: Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. What do you mean by group? (I can think of two different answers depending on what you mean. either group= in chan_dahdi.conf or [trunkgroups] in the same file. Both are completely different) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Are you trying to do NFAS, only have one D chan for two PRIs? It works very well, just search around for NFAS examples on the web or voip-info.org. It saves you a channel and in some cases as much as $100/mo per D chan you consolidate. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with transfer application (REFER)
Hi nik600, I had some trouble transferring calls with that version of Asterisk even if I used the normal transfer via features.conf. Upgrading to 1.4.24 helped a bit (even if not completely). My advice is to upgrade to 1.4.24 or the latest. Giorgio nik600 wrote: I'm experiencing some problem using the transfer() application,expecially when a call in received from a queue. I'm using Asterisk 1.4.22.1 This is my scenario: ; this is the piece of code in extensions.conf that place the call in the queue when is called exten = ,1,Answer exten = ,n,Queue(2000|t) ;this is the piece of code that calls the user test when is called exten = ,1,Dial(SIP/test) ; this is the piece of code that transfer the call using REFER exten = ,1,Transfer(SIP/endpo...@x.y.z.t) Calling the call is placed on the queue, and then answered from a member (SIP/test), when the member try to transfer the call to the call ends with an error every time. Calling the call is placed directly to the user SIP/test, when the user try to transfer the call to SOMETIMES the call ends with an error. Sometimes asterisk says: Auto fallthrough, channel 'SIP/xx' status is 'ANSWER' and sometimes it says Auto fallthrough, channel 'SIP/xx' status is 'UNKNOWN' Can you help me to guess the problem? I've read that the REFER implementation in the transfer application is not complete, is it true? Is there any procedure / configuration to use a complete and stable implementation of the REFER functionality? Thanks to all in advance -- Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com vo...@work - The Agile PBX http://www.voiceatwork.eu FGA srl - http://www.fgasoftware.com Tel: 02 997663.14, Fax: 02 91390172 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Calling Feature?
On Fri, Jun 12, 2009 at 8:43 AM, Christopher Stamper christopherstam...@gmail.com wrote: On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote: Nerdvittles.com has a nice example of this, when they are up. They used it for Phone trees for a school or something like that. Took less than 30 minutes to put in my dialplan and use Sounds like exactly what I am looking for... I went to nerdvittles.com and searched, but couldn't find it. Spoke too soon, I just found what I think you were referring to: TeleYapper. Looks excellent! Thanks! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Calling Feature?
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote: Nerdvittles.com has a nice example of this, when they are up. They used it for Phone trees for a school or something like that. Took less than 30 minutes to put in my dialplan and use Sounds like exactly what I am looking for... I went to nerdvittles.com and searched, but couldn't find it. you can have a script generate a list of call files which automatically ring the callers in the list and play a message I may have to end up doing that. Problem is, the people who will be recording the message want it to be really easy; like, call a number, talk and hang up. I guess I could do that, but it may end up becoming a huge project... I was hoping that someone already had done this. Thanks for the suggestions, I'll keep looking! -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current possible values for DIALSTATUS?
According to app_dial.c these are the present values (1.4.25-rc2; I assume these are the same for 1.6.1.1) DIALSTATUS - This is the status of the call:\n CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n DONTCALL | TORTURE | INVALIDARGS\n _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Thursday, June 11, 2009 8:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Current possible values for DIALSTATUS? Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help building dahdi for debian
Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current possible values for DIALSTATUS?
Danny Nicholas schrieb: According to app_dial.c these are the present values (1.4.25-rc2; I assume these are the same for 1.6.1.1) DIALSTATUS - This is the status of the call:\n CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n DONTCALL | TORTURE | INVALIDARGS\n core show application Dial _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Who says DIALSTATUS can ever be FAIL or FAILED? If that actually happens then that would be a bug in the documentation. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Fri, 2009-06-12 at 23:58 +1000, Alex Samad wrote: what is the best way forward to recompile with hardware echo canceller support. No need to do anything special during compilation. For hardware echo cancellation just put the option echocancel=yes in chan_dahdi.conf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on static IP and Sipura-1001 on dynamic IP (was: Re: asterisk-users Digest, Vol 59, Issue 28)
Kengie Ho schrieb: I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Whenever the ATA gets a different IP address from a DHCP server (?) it should re-register automatically. If the Sipura has a configuration parameter to control the registration time then try setting that to a few minutes. More often then not the default is an hour or more. If it does not have such a parameter then you can try setting something like in following in Asterisk's sip.conf: minexpiry=65 defaultexpiry=145 maxexpiry=185 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote: Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi What do you mean? I suspect that the following patch: http://patch-tracking.debian.net/patch/series/view/dahdi-linux/1:2.2.0~dfsg~rc5-1/no_firmware_download BTW: legal purity aside, downloading an external source at build time is generally a big no-no for a build server on Debian. Thus this downloading breaks my intention to get the modules distributed in the distribution as part of linux-modules-2.6 package (which includes all sorts of external modules). If you're not using those packages (and build at a place with internet connectivity) you should have no problem. Which is the case for you? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current possible values for DIALSTATUS?
Thanks for the reply. I may be mistaken for assuming that 'disposition' values in CDR were actually from DIALSTATUS. My CDR table has entries for 'disposition' including ANSWERED, FAILED, NO ANSWER. JR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Current possible values for DIALSTATUS? Danny Nicholas schrieb: According to app_dial.c these are the present values (1.4.25-rc2; I assume these are the same for 1.6.1.1) DIALSTATUS - This is the status of the call:\n CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n DONTCALL | TORTURE | INVALIDARGS\n core show application Dial _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Who says DIALSTATUS can ever be FAIL or FAILED? If that actually happens then that would be a bug in the documentation. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending sip info messages
Hi all, I`m searching for a special solution to send text messages inside Sip info packets, that are normally used for dtmf signalization. So far I’m able to exchange sip Info messages between two softphones which are connected directly together (only over a Switch). By connecting both Softphones on the asterisk pbx, registration is ok and the voice interconnection is also fine. During the call, when I start sending a sip info message to my remote station, first the asterisk receives it and then sends back a sip status message: Status: 403 Unauthorized Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGER, SUSCRIBE, NOTIFY X-Asterisk-HangupCause: normal Clearing Asterisk doesn’t relay the info message to my remote station. In the sip.conf file I changed the dtmfmode from “auto” to “info” but without any improvements. In the “Allow:…” line of the received status message there is no “INFO” entry. Perhaps could this be the reason why asterisk don’t forwards the info message to my remote station, and if so, how can I change asterisk configuration to allow info messages? Thank you very much for some help! Regards Karsten -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current possible values for DIALSTATUS?
IMO, the disposition in the CDR is set at hangup/fail time, not dial time. I'm sure many others can elaborate. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, June 12, 2009 10:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Current possible values for DIALSTATUS? Thanks for the reply. I may be mistaken for assuming that 'disposition' values in CDR were actually from DIALSTATUS. My CDR table has entries for 'disposition' including ANSWERED, FAILED, NO ANSWER. JR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Current possible values for DIALSTATUS? Danny Nicholas schrieb: According to app_dial.c these are the present values (1.4.25-rc2; I assume these are the same for 1.6.1.1) DIALSTATUS - This is the status of the call:\n CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n DONTCALL | TORTURE | INVALIDARGS\n core show application Dial _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Who says DIALSTATUS can ever be FAIL or FAILED? If that actually happens then that would be a bug in the documentation. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current possible values for DIALSTATUS?
John Regal schrieb: I may be mistaken for assuming that 'disposition' values in CDR were actually from DIALSTATUS. My CDR table has entries for 'disposition' including ANSWERED, FAILED, NO ANSWER. True. and BUSY. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Danny Nicholas schrieb: According to app_dial.c these are the present values (1.4.25-rc2; I assume these are the same for 1.6.1.1) DIALSTATUS - This is the status of the call:\n CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n DONTCALL | TORTURE | INVALIDARGS\n core show application Dial _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Who says DIALSTATUS can ever be FAIL or FAILED? If that actually happens then that would be a bug in the documentation. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + TC400B - Clock Trouble
Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call, it takes 25 seconds for the voice to go to the other end... I have Asterisk 1.4.25.1, Dahdi 2.2.0-rc5 on a CentOS-5.3 (x86_64) server with a 2.6.18-128.1.10.el5 linux kernel Ast CLI when calling with g729 ast-01*CLI transcoder show 1/1 encoders/decoders of 92 channels are in use. Dahdi start returns: (SCREEN):r...@ast-01:[~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: wctc4xxp:[ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: [ OK ] DMESG returns: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0-rc5 dahdi_transcode: Loaded. wctc4xxp: tc400b0: Attached to device at :0f:03.0. wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M dahdi_transcode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) dahdi_transcode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) dahdi: Registered tone zone 30 (Switzerland) -- -- Marc LEURENT ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sip info messages
Asterisk as of this writing doesn't really support text messaging except for asterisk-to-phone during a call. Some folks on this thread use other programs in conjunction with asterisk to support this capability. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Schubotz Sent: Friday, June 12, 2009 10:16 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sending sip info messages Hi all, I`m searching for a special solution to send text messages inside Sip info packets, that are normally used for dtmf signalization. So far Im able to exchange sip Info messages between two softphones which are connected directly together (only over a Switch). By connecting both Softphones on the asterisk pbx, registration is ok and the voice interconnection is also fine. During the call, when I start sending a sip info message to my remote station, first the asterisk receives it and then sends back a sip status message: Status: 403 Unauthorized Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGER, SUSCRIBE, NOTIFY X-Asterisk-HangupCause: normal Clearing Asterisk doesnt relay the info message to my remote station. In the sip.conf file I changed the dtmfmode from auto to info but without any improvements. In the Allow: line of the received status message there is no INFO entry. Perhaps could this be the reason why asterisk dont forwards the info message to my remote station, and if so, how can I change asterisk configuration to allow info messages? Thank you very much for some help! Regards Karsten -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, once again - last time to publish this.. i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) I would be glad if someone can take a look... Kindly regards, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple PRI's in one group ..how??
I mean..making a single trunk between a pstn or telco with 2 or more PRI's.. I mean instead of using 32 channels to use 64 or more.. I am trying to increase the capacity between my PSTN and asterisk actually. There will be more than 35-40 concurrent calls so while creating a zap trunk(or dahdi..whatever u call) i want all pris to behave like they are a single span I hope i make myself clear now :) sorry for misunderstanding. On Fri, Jun 12, 2009 at 8:13 AM, Tzafrir Cohentzafrir.co...@xorcom.com wrote: On Fri, Jun 12, 2009 at 11:49:02AM +0300, Oguzhan Kayhan wrote: Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. What do you mean by group? (I can think of two different answers depending on what you mean. either group= in chan_dahdi.conf or [trunkgroups] in the same file. Both are completely different) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Are you trying to do NFAS, only have one D chan for two PRIs? It works very well, just search around for NFAS examples on the web or voip-info.org. It saves you a channel and in some cases as much as $100/mo per D chan you consolidate. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Stefan Agethen schrieb: Hey Everyone, once again - last time to publish this.. Hey, you posted this on Jun 8, 10 and 12. i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. Which firmware version is on the phones? I can't remember having seen any problems on the Snoms related to putting calls on hold. You could try to post on http://groups.google.de/group/gemeinschaft-users (gemeinschaft-us...@googlegroups.com) even if you're not using Gemeinschaft. Maybe somebody happens to know why that doesn't work. Just make sure to say Ich verwende zwar nicht Gemeinschaft, aber vielleicht hat trotzdem jemand zufällig einen Tipp für mich. And I'd start the subject with OT (off-topic). In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the snoms doesnt understand this and thinks the caller is still on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --ASTERISK IS INVITING MY SNOM AT HOME-- Audio is at [ I P - A S T E R I S K ] port 11576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40 (slin) to SDP Adding non-codec 0x1 (telephone-event) to SDP --SNOM IS ANSWERING THE CALL-- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port [ I P - A N G E R U F E N E R ]:13790 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The same as above.. --NOW I PRESS HOLD ON THE SNOM, THE SIP STATEMENT TO THE ASTERISK IS-- --- SIP read from [ I P - A N G E R U F E N E R ]:5060 --- BYE sip:[ TEL. CALLER ]...@[ I P - A S T E R I S K ] SIP/2.0 Via: SIP/2.0/UDP [ I P - A N G E R U F E N E R ]:5060;branch=z9hG4bK-o6cb4olp9iv1;rport From: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;tag=e8yr1936gy To: [ MyName in the Snom ], [ MyName in the Snom ], ;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89e...@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: sip:4...@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-debug-patton-snom.txt ) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AmooCon video recordings online
JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple PRI's in one group ..how??
Oguzhan Kayhan escribió: I mean..making a single trunk between a pstn or telco with 2 or more PRI's.. I mean instead of using 32 channels to use 64 or more.. I am trying to increase the capacity between my PSTN and asterisk actually. There will be more than 35-40 concurrent calls so while creating a zap trunk(or dahdi..whatever u call) i want all pris to behave like they are a single span I hope i make myself clear now :) sorry for misunderstanding. Just put all channels of all spans you want on the same group (look at this example snip of zapata.conf): group=1 signalling = pri_cpe context=4pri channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 That way asterisk will use the 120 channels as one big trunk on g1. if you Dial(Zap/g1/number), it will use all the channels. If you want to do round-robin so it goes around all channels and not the first free, try Dial(Zap/r1/number) from incrementing round-robin, or Dial(Zap/R1/number) to decrementing round-robin. Of course, depending on your need you can split them off on the arrange you want, not only at a entire PRI level. If you wanted to reserve 5 channels to inbound calls only and the rest for outbound, you could do this for example: group=1 signalling = pri_cpe context=outbound-group channel = 6-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 group=7 signalling = pri_cpe context=inbound-group channel = 1-5 This way, if you Dial group 1, you won't use channels 1-5, leaving them free for inbound calls. Cheers, Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Fri, Jun 12, 2009 at 05:40:16PM +0300, Tzafrir Cohen wrote: On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote: Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi What do you mean? I suspect that the following patch: http://patch-tracking.debian.net/patch/series/view/dahdi-linux/1:2.2.0~dfsg~rc5-1/no_firmware_download yeah I downloaded the source and found this patch, BTW: legal purity aside, downloading an external source at build time is generally a big no-no for a build server on Debian. Thus this downloading breaks my intention to get the modules distributed in the distribution as part of linux-modules-2.6 package (which includes all sorts of external modules). I understand, but it was feasible in zaptel to build a package were all you had to do was download the firmware - I haven't looked at the dep / source requirements so I am not sure if the is feasible on the debian build servers If you're not using those packages (and build at a place with internet connectivity) you should have no problem. Which is the case for you? building my own package from your source just removing the above offending patch :) any chance of getting digium to host a digium debian repo (sort of how virtulbox doit), that way they could have a fully build package ? alex -- Neither in French nor in English nor in Mexican. - George W. Bush 04/21/2001 declining to take reporters' questions during a photo op with Canadian Prime Minister Jean Chretien signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AmooCon video recordings online
No divx hd? just kidding OT: Odd how many video/audio standards there are, and the growing issue with them? I recall when you had two choice Windows Media or RealPlayer. Now I have to make 3-4 for everything from DivX to iPod to Walkman. For example my cell phone can't play a H264/AAC due the cpu requirements needsI'm happy to see Windows 7 added H264/AAC natively... Imagine adding 5-6 more audio codecs and having to support them...like we don't have enough already... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 3:49 PM To: Asterisk Users Subject: [asterisk-users] AmooCon video recordings online JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple PRI's in one group ..how??
2009/6/12 Miguel Molina mmol...@millenium.com.co: Oguzhan Kayhan escribió: I mean..making a single trunk between a pstn or telco with 2 or more PRI's.. I mean instead of using 32 channels to use 64 or more.. I am trying to increase the capacity between my PSTN and asterisk actually. There will be more than 35-40 concurrent calls so while creating a zap trunk(or dahdi..whatever u call) i want all pris to behave like they are a single span I hope i make myself clear now :) sorry for misunderstanding. Just put all channels of all spans you want on the same group (look at this example snip of zapata.conf): group=1 signalling = pri_cpe context=4pri channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 That way asterisk will use the 120 channels as one big trunk on g1. if you Dial(Zap/g1/number), it will use all the channels. If you want to do round-robin so it goes around all channels and not the first free, try Dial(Zap/r1/number) from incrementing round-robin, or Dial(Zap/R1/number) to decrementing round-robin. Of course, depending on your need you can split them off on the arrange you want, not only at a entire PRI level. If you wanted to reserve 5 channels to inbound calls only and the rest for outbound, you could do this for example: group=1 signalling = pri_cpe context=outbound-group channel = 6-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 group=7 signalling = pri_cpe context=inbound-group channel = 1-5 This way, if you Dial group 1, you won't use channels 1-5, leaving them free for inbound calls. Cheers, Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center You can also gain 3 voice channels and potentially save money by using NFAS which Asterisk supports beautifully. In one instance, I was able to save $2,100 month by using seven D chans on a T3 (DS3) rather than twenty eight. GXing wanted $100/mo per D chan. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AmooCon video recordings online
Jason Aarons (US) schrieb: OT: Odd how many video/audio standards there are, and the growing issue with them? I recall when you had two choice Windows Media or RealPlayer. There is only one format[1] of choice: .mov :-) It's amazing how formats natively supported by QuickTime play smoothly at high resolution even with 2 virtual machines and all sorts of other stuff running on my MacBook. That's next to impossible with .wmv/.flv videos and Flip2Mac / Perian. divx kinda works but requires an additional player. Apple must have put an incredible amount of work into QuickTime optimizations. Now I have to make 3-4 for everything from DivX to iPod to Walkman. For example my cell phone can't play a H264/AAC due the cpu requirements needsI'm happy to see Windows 7 added H264/AAC natively... [1] Mixing up container formats and codecs here. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AmooCon video recordings online
I just wish my HTC Touch Pro cell phone or my PlayStation3 could play .mov -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AmooCon video recordings online Jason Aarons (US) schrieb: OT: Odd how many video/audio standards there are, and the growing issue with them? I recall when you had two choice Windows Media or RealPlayer. There is only one format[1] of choice: .mov :-) It's amazing how formats natively supported by QuickTime play smoothly at high resolution even with 2 virtual machines and all sorts of other stuff running on my MacBook. That's next to impossible with .wmv/.flv videos and Flip2Mac / Perian. divx kinda works but requires an additional player. Apple must have put an incredible amount of work into QuickTime optimizations. Now I have to make 3-4 for everything from DivX to iPod to Walkman. For example my cell phone can't play a H264/AAC due the cpu requirements needsI'm happy to see Windows 7 added H264/AAC natively... [1] Mixing up container formats and codecs here. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote: any chance of getting digium to host a digium debian repo (sort of how virtulbox doit), that way they could have a fully build package ? Or resolve the issues that made this patch necessary in the first place. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Sat, Jun 13, 2009 at 01:40:48AM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote: any chance of getting digium to host a digium debian repo (sort of how virtulbox doit), that way they could have a fully build package ? Or resolve the issues that made this patch necessary in the first place. To get this to work can i simply apt-get source dahdi-linux modify debian/patches/series to comment out no_firmware_download then dpkg-buildpackage -rfakeroot -us -uc -b should that work ? -- I was not prepared to shoot my eardrum out with a shotgun in order to get a deferment. Nor was I willing to go to Canada. So I chose to better myself by learning how to fly airplanes. - George W. Bush 02/25/1990 Dallas Morning News signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fedora Core 10 and g729 codec
Hi All; Did anyone tried Fedora Core 10 with Asterisk 1.4.25.1? I am facing a problem that it is not able to detect the g729 (although I have another machines running fedora core 9 and it is fine). Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + TC400B - Clock Trouble
I'm not sure if the kernel timing HZ has anything to still do with things anymore. You might need to recompile your kernel with HZ=1000 -Jon lf...@leurent.eu wrote: Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call, it takes 25 seconds for the voice to go to the other end... I have Asterisk 1.4.25.1, Dahdi 2.2.0-rc5 on a CentOS-5.3 (x86_64) server with a 2.6.18-128.1.10.el5 linux kernel _Ast CLI when calling with g729_ ast-01*CLI transcoder show 1/1 encoders/decoders of 92 channels are in use. _Dahdi start returns:_ (SCREEN):r...@ast-01:[~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: wctc4xxp:[ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: [ OK ] _DMESG returns:_ dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0-rc5 dahdi_transcode: Loaded. wctc4xxp: tc400b0: Attached to device at :0f:03.0. wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M dahdi_transcode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) dahdi_transcode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) dahdi: Registered tone zone 30 (Switzerland) -- -- Marc LEURENT -- Scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content by MailScanner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0.10: core restart on ReceiveFax()
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
exten = 8447,1,Answer()
exten = 8447,n,GoSub(Capture-Fax,s,1)
exten =_NXXNXX,1,Answer()
exten =_NXXNXX,n,GoSub(DialOut-PSTN,s,1(1${EXTEN}))
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN}))
exten =_91NXXNXX,1,Answer()
exten =_91NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN:1}))
Actual outbound faxes work correctly. That is, a call from 173 to an
outside fax machine works.
The test faxes go to:
[Capture-Fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif) ;; 1.6 use ReceiveFAX
exten = s,n,Hangup()
When the test fax gets to ReceiveFax() asterisk restarts. Any calls at
the time are lost.
-- Executing [8...@outbound-fax:1] Answer(SIP/173-081d3780, )
in new stack
-- Executing [8...@outbound-fax:2] Gosub(SIP/173-081d3780,
Capture-Fax,s,1) in new stack
-- Executing [...@capture-fax:1] Set(SIP/173-081d3780,
FAXFILE=/var/spool/asterisk/fax/20090612_1710) in new stack
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-081d3780,
/var/spool/asterisk/fax/20090612_1710.tif) in new stack
/var/spool/asterisk/fax exists, permissions 777:
ls -l /var/spool/asterisk
total 32
..
drwxrwxrwx 2 root root 4096 2009-05-03 14:21 fax
...
I've set debug and verbose to 20, but no further info.
What am I missing? Anybody have something like this working this working?
sean
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
