Re: [asterisk-users] Problems with audio
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias wrote: > Yes my friend...CONFIRMED!!! G729 on both sides > > If the problem happen with SIP to SIP calls and with the same codec, the problem is inside the phone. Check if you can pump up the volume inside his configuration. What phones are you using? -- -- Adrià Vidal -- _ -- 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 Calendar
On Wed, Sep 8, 2010 at 4:24 PM, Adrià Vidal wrote: Sorry was my fault , res_calendar was ok, but ical and caldav need other libs (neon,ical...) that were not installed in my system. -- -- Adrià Vidal -- _ -- 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 Calendar
On Wed, Sep 8, 2010 at 3:56 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Adrià Vidal > *Subject:* [asterisk-users] asterisk 1.8 Calendar > > > > >I'm testing some of the new features of Asterisk 1.8, > >but seems an impossible mission to make the calendars run. > > >cli show empty: > > >**CLI> calendar show > calendars > > >Calendar Type > Status > > > -- * > > >And i can't see anything into the log, calendar.conf is ignored. > > >Any sugestion? > >Adrià Vidal > > Start here > > https://issues.asterisk.org/bug_view_page.php?bug_id=14771 > > > Nothing related to my problem there, will post a bug if anyone having the same problem here. -- -- Adrià Vidal -- _ -- 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 1.8 Calendar
I'm testing some of the new features of Asterisk 1.8,
but seems an impossible mission to make the calendars run.
cli show empty:
**CLI> calendar show
calendars
Calendar Type
Status
-- *
And i can't see anything into the log, calendar.conf is ignored.
Any sugestion?
--
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
try to have a dns cache into your LAN, Aastra phone are prone to fail when have any little DNS error. -- -- Adrià Vidal -- _ -- 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] Fw: OfficeSIP Softphone
On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine wrote: > Hello, > > Yes, unfortunately, the sip client lib does not support udp. > > Best regards, > Vitali Fomine > > > Then check you are using an Asterisk patched for TCP. -- -- Adrià Vidal -- _ -- 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] Fw: OfficeSIP Softphone
You are running an Asterisk version for SIP TCP ? your SIP UA seems talking SIP over TCP Via: SIP/2.0/TCP 192.168.1.15:56298 Max-Forwards: 70 From: ;tag=2baacde98c;epid=aa3c1b27a7 To: Call-ID: 28a90e7402da49159f343be9bc82b4d0 CSeq: 1 SERVICE Contact: ;proxy=replace;+sip.instance="" User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:mrasloc.trixbox1.lo...@trixbox1.local", nonce="36662fdf", response="d6f90f263010891a42b3f7d46113796a" Content-Type: application/msrtc-media-relay-auth+xml Content-Length: 395 -- -- Adrià Vidal -- _ -- 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 Aler-info Ringtone
Have someone running fine Alert-Info with a Snom 370
( System Information:
Phone Type: snom370-SIP
MAC-Address:0004132661BD
IP-Address: 192.168.10.170
Firmware-Version: snom370-SIP 7.3.14 14961)
i've tried
exten =>
200,1,SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-external)
exten => 200,n,Dial(SIP/${EXTEN},30)
Can see into the phone SIP trace is receivend the Alert-Info, but
phone continue to ring with the default one.
"SIP Trace"
INVITE sip:2...@192.168.10.170:1110 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.5:5060;branch=z9hG4bK29710073;rport
From: "Ilimit " ;tag=as21549e4b
To:
Contact:
Call-ID: 4e0f770e17b79d137f59575a43af8...@192.168.10.5
CSeq: 102 INVITE
User-Agent: Asterix PBX
Max-Forwards: 70
Date: Thu, 26 Feb 2009 17:00:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Alert-Info:alert-external
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 15980 15980 IN IP4 192.168.10.5
s=session
c=IN IP4 192.168.10.5
t=0 0
m=audio 19616 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Any suggestion?
thanks
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adriavi...@gmail.com
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Re: [asterisk-users] Asterisk - Trixbox
On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett wrote: > My provider migrated from an old EOL softswitch to Trixbox. > > I have a number (8159093011) on a different server on a different network. > It appears as though the incoming calls are trying to authenticate against > that number, which isn't present on the box. Could someone help me decode > this debugging output? I was calling 8159911010. My server is > 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other > server by adding insecure settings, but that didn't seem to solve it on this > one. > > http://pastebin.com/f5151341f > > > - > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten => s,1,NoOP(-incoming call---) exten => s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- 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] problems of DNS
On Tue, Dec 16, 2008 at 11:04 PM, troxlinux wrote: > Hi list, I have for a year I have an account to call with broadvoice from > about 3 days beginning a not registered problem of, asterisk shows to a > message of error with the DNS, and my dns this working fine > > WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for > registration to 908...@sip.broadvoice.com@sip.broadvoice.com, trying > REGISTER again (after 20 seconds) > [Dec 16 16:00:18] NOTICE[5770]: chan_sip.c:7517 sip_reg_timeout:-- > Registration for '908xxx...@sip.broadvoice.com@sip.broadvoice.com' timed > out, trying again (Attempt #70) > These line don't look very well Registration for '908xxx...@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #70) check the register line. -- -- Adrià Vidal adriavi...@gmail.com ___ -- 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] Calls randomly being placed on hold...
Enable dtmf on the logger.conf and see if you get some # or ** or whatever key you have configured at features.conf for transfer, maybe you could see something into the logs. I get something similar with some Linksys PAP2. Adria Vidal On Tue, Apr 1, 2008 at 10:15 PM, Tim Nelson <[EMAIL PROTECTED]> wrote: > Hello! I'm having a bit of an issue with one of my installations that I > cannot figure out. For some reason, when two people are in a call (both > local to the * box, same subnet, pure SIP), the call will randomly be placed > on hold and provide MOH to the other party. We're using Polycom IP430 > handsets almost exclusively for this installation. Can anyone think of a > reason why a call would randomly go on hold? > > Tim Nelson > Systems/Network Support > Rockbochs Inc. > > > ___ > -- 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 > -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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] Got SIP response 406 "Not Acceptable"
try doing a sip debug peer XXX (the problematic extension) and then send a call to him till fail, then see the log, or send a piece here. On Thu, Mar 27, 2008 at 3:10 AM, Al lists <[EMAIL PROTECTED]> wrote: > Nope, > Coded is Ulaw on both sides and also this issue happens occasionally with > no change. > > > > On Wed, Mar 26, 2008 at 6:17 PM, Adrià Vidal <[EMAIL PROTECTED]> wrote: > > > Seems a codec problem, check the sip.conf from that spa942 > > > > On Wed, Mar 26, 2008 at 11:59 PM, Al lists <[EMAIL PROTECTED]> wrote: > > > > > I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" > > > occasionally when try to dial to SPA942 , > > > anyone has any idea on this before i consider Firmware upgrade? > > > > > > > > > ___ > > > -- 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 > > > > > > > > > > > -- > > -- > > Adrià Vidal > > [EMAIL PROTECTED] > > ___ > > -- 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 > -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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] Got SIP response 406 "Not Acceptable"
Seems a codec problem, check the sip.conf from that spa942 On Wed, Mar 26, 2008 at 11:59 PM, Al lists <[EMAIL PROTECTED]> wrote: > I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" > occasionally when try to dial to SPA942 , > anyone has any idea on this before i consider Firmware upgrade? > > > ___ > -- 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 > -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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] (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released
Isasterisk-1.4-current.tar.gz(13-Mar-2008 15:06 11M) not the same as asterisk-1.4.18.1.tar.gz (18-Mar-2008 12:24 11M ) ? Should be? ___ -- 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 asterisk and aastra phones
Ours have been running fine since pointing the aastra.cfg to the LAN NTP. Don't know what can be happening with yours. On Tue, Feb 26, 2008 at 12:52 AM, Marius Muja <[EMAIL PROTECTED]> wrote: > There already is an ntp server on the LAN, but the phones still freeze. > > On Mon, Feb 25, 2008 at 2:18 PM, Adrià Vidal <[EMAIL PROTECTED]> wrote: > > > Aastra tech are a bit slow, be sure to put a ntp server into your LAN > > and point Aastra's to it. > > Your problems will be solved. > > > > Adrià Vidal > > > > > > ___ > > -- 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 > -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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 asterisk and aastra phones
Aastra tech are a bit slow, be sure to put a ntp server into your LAN and point Aastra's to it. Your problems will be solved. Adrià Vidal ___ -- 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] Best "Console" phone?
Actualy Aastra phone are limited to control "only" 50 BLF, snom360 can handle in our site about 110, and only seems a bit busy time to time. adrià vidal ___ -- 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 DIAL() premature timeout on a PRI trunk to legacy PBX
> > > > Set callprogress=no and busydetect=no in > > Worked for me, thanks a lot!!! -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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 desktop tools for OS X
I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee <[EMAIL PROTECTED]> wrote: > Hi everyone, > > I have been long working on a project (http://asterisktools.org, to be > released under GPL) that aims to provide desktop tools for Macs. I am > finally getting to the release stages of this application and hope to > have an early BETA available next weekend. > > If there is anybody who is interested in this tool, please send me an > email as I am looking for people who can test the application for me > before we make a final release. > > The code is already available via SVN and there are some really cool > and thoughtful features. > > Thanks a lot. > > -- > "I never look back darling, it distracts from the now", Edna Mode (The > Incredibles) > > ___ > -- 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 > -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- 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] Cisco Buddies
O.K looks like you are talking about "presence", take a look about the hint app. But didn't now how to make the Cisco check the hint... get us informed about you advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Buddies
2006/8/10, Peder @ NetworkOblivion <[EMAIL PROTECTED]>: Can you do buddies with Cisco phones running SIP? I can't find anything that says yes or no. I can set it up on the * server, but I don't know what to do on the 7960's themselves. What about a google look for asterisk cisco 7960 config in google? Firts and second looks great. Cisco 7960 IP Phone - SIP configuration - [ Traduzca esta página ] If you want to know how to configure your Cisco 7960 IP Phone to work with the skinny protocol (SCCP) and Asterisk PBX just click here. User Comments ... www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html - 33k - En caché - Páginas similares Asterisk phone cisco 79xx - voip-info.org - [ Traduzca esta página ] Asterisk Cisco 79XX XML Ser... Asterisk config sip.conf, Asterisk Linksys NSLU2, Cisco 7940-7960 auto-answer... cisco 79xx, Cisco Phones, Codecs ... www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx - 76k - -- Adrià Vidal [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip incoming stop working, what to look for in logs?
try a: Sip debug and see what comes into CLI when the incoming call is rejected. (maybe changed something at your contexts or sip.conf? ) adrià ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 headset with static noise
2006/7/17, Christian Stredicke <[EMAIL PROTECTED]>: There is a difference in the biasing circuit for the microphones in the headsets. Unfortunately there is no standard on the market. The snom phones 190/320/360 (let's say: type A) behave different than snom 300 (type B). So there is always the need to have different headsets or different cables (Quick Disconnect). Some headsets are just working with one type (those with extra amplifier) and other devices seem to work in both environments, but that's not really true. The headsets are always working much better with just one type. So if someone has a headset designed for type A, he'll have a bad quality while connecting it to type B phones although he is able to here something. Don't forget to have a connection to an earth-signal (e. g. shielded Ethernet cable to PC/switch or earth-grounded power supply). Hope this helps, CSWe are using the Snom 300 with Snom HS MM headsets,(yes the switch and cable are ground-signaled) all bought together. We are testing it for a call center (100 seats) deploy and as i said first we are very disappointed with the performace of them. Going to RMA them and get some Linksys 841. The old product from Snom (360) runned headsed really well. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 headset with static noise
Have a look at this document: http://www.snom.com/wiki/index.php/FAQs#Q:_Why_is_there_a_humming_noise_when_using_the_headset.3FMichielThanks Michiel, that was the second thing i do, phone was connected to a well powered/connected switch.I could understand a chep headset would do that, but a 30 euro headset maybe is not goingto be the best... but should perform quite better.Maybe is a design fault , because the same headset connected into another voIP phone runs fine. Snom maybe have gone too cheap and too bad for his snom 300 ? -- Adrià Vidal[EMAIL PROTECTED] | [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 300 headset with static noise
Someone using these phone Snom 300 with his own headset ?We got horrible static noise on them?P.D.Got silence as answer from Snom by now... maybe on holidays or with in theEuropean Football championship. -- Adrià Vidaladriavidal at gmail.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX + Digium + SpanDSP
2006/6/16, Boris Bakchiev <[EMAIL PROTECTED]>: Hi, We do J We use iaxmodem+hylafax combo on TE406P card. Around 4K of faxes were received without any problems (some faxes are over 80 pages long!) It is working really well! Have Faxdispatch running well with Hylafax? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spanDSP & app_rxfax.so
Running asterisk-1.2.7.1 & spandsp-0.0.2pre25.tar.gz Have compiled spandsp,but patch give me one warning Asterisk (pbxdev:/usr/src/asterisk-1.2.7.1/apps# patch <../apps_Makefile.patch patching file Makefile Hunk #2 FAILED at 104. 1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Can do a make and copy by hand the module, but get: pbxdev*CLI> load app_rxfax.so Unable to load module app_rxfax.so May 24 10:42:41 WARNING[1339]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident Any suggestion? -- Adrià Vidal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP register problem
You changed your default SIP (bindport) port to 5061 at the server, so your client needs to look there. Try like these register => sipteszt:[EMAIL PROTECTED]:50/sipteszt bindport=5061 ; UDP Port to bind to (SIP standard port is 5060) Adrià Vidal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 License questions
When your phones acces to voicemail or to an IVR into the asterisk then a G729 license is used so Asterisk is transcoding. So you are gonna use the licenses for sure, maybe not from phone to phone calls, but yes using the Asterisk functions. Adrià Vidal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy
Someone have good sound on ChanSpy with SIP channelsa at an Asterisk 1.2.4 ?Mine is cracking all the time.-- Adrià Vidal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I see Asterisk 1.2.2 into the ftp or was a vision?
Someone can confirm the new release is out? Haven's seen any post about it! -- Adrià Vidal [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'ztmonitor' stopped working after using 'fxotune'
2005/11/18, Chuck Bunn <[EMAIL PROTECTED]>: > Hi, > > I cannot get 'ztmonitor' to run anymore after I ran 'fxotune'. I get the > following error: > > [EMAIL PROTECTED] ~]# cd /usr/src/zaptel > [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 - > Unable to open /dev/dsp: No such file or directory > Cannot open audio ... > [EMAIL PROTECTED] zaptel]# Try: ./ztmonitor 1 -v Adrià Vidal ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 3 + AVM Fritz ?
Someone have info about install an AVM fritz into FC3 ? I'm getting problems with kernelcapi, after succesfully installed the fcpci support. Thanks -- Adrià Vidal [EMAIL PROTECTED] Mail is better with 1Gmail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Having Trouble Creating an IVR
Maybe you are using a Sipura ATA? they use *XX to acticate/deactivate advanced functions (DND,Call Return Code:Blind Transfer Code,Call Back Act Code,DND Act Code) or Block ANC Act Code:*77 2005/7/12, Tim P <[EMAIL PROTECTED]>: > I have asterisk 1.0.5 installed via apt on a debian system. It's a > custom distrobution called Voyage Linux that runs from a flash card > and I have a hard drive installed with mysql installed on it as well > as apache. I have been though the AMP install guide (asterisk > management portal) and in the interface it has a place for me to > record new IVR menus. I have to dial *77 to begin recording but it > just gives me a beep-beep-beep like I have an invalid line when > attemping to dial it. *77 exists in my extensions.conf and I am able > to make and recieve calls from the extension I am using. I didn't > think this sounded like an AMP problem as the conf files has the > entries in there. > > I looked in the astrisk "full" logfile but saw no errors pertaining to > this nor did I see anything in the console when I connected with > asterisk -rv and did a sip debug ip 192.168.5.114:5060 (my > extensions's ip and port). > > Where should I begin with troubleshooting? > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Adrià Vidal [EMAIL PROTECTED] Mail is better with 1Gmail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap gives no ring to the caller...
On Mon, 29 Nov 2004 20:38:02 -0500, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On November 29, 2004 08:22 pm, [EMAIL PROTECTED] wrote: > > this finally do the trick, thanks to everybody > > > > exten => 97XX,1,Answer > > exten => 97XX,2,Ringing > > exten => 97XX,3,Wait(5) > > exten => 97XX,4,Goto(menuCARS,s,1) > > Out of curiosity why would you Answer on a CAS or PRI E1 until you were ready > to answer? And you do have a _ in front of that 97XX right? :-) > > -A. I've tried many things, and Answer was one more. None _ in front of the number (i've put X only to hide the real numbrer, sorry) -- Adrià Vidal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
