[asterisk-users] leading ghost 0: SOLVED
Hi all, the problem has been solved setting pridialplan=unknown. Asterisk restart is mandatoryreloading chan_dahdi.conf module is not enough! The leading zero still remains a ghost since it cannot be seen in any log. B! Creepy! :) Hope this can help some other soul in pain... Thank you all for your answers, guys! Giorgio Incantalupo -- _ -- 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] chan_capi audio quality issue
Hi everyone! I'm installing in our small office a phone system using a Fritz!Card USB that I found in my attic. I'm using Asterisk 1.8.13, FreePBX 2.11, chan_capi 1.1.6. Everything works great, I can place outgoing calls, ingoing calls work, internal calls too, voicemail, ... The only problem is the audio quality when using chan_capi to place calls. The remote party can not hear me clearly and from the asterisk side, I hear a clicking noise (I can try to record it if that helps) repeating at close intervals but only when someone is speaking or when there's music (basically, when there is no silence). I've skimmed Google for answers and came up with some old posts dating back to 2003/4/6/... speaking about the CAPI_MAX_B3_BLOCK_SIZE value in chan_capi.h. I tried to tweak that value and recompile several times. Each time, I have the same clicking noise but the interval changes. I can't get rid of the noise though. This is currently preventing me from putting my setup in production and I'd really like to find a solution for this. I'll be glad to hear any pointers you can throw at me. I can basically try everything as this is not in prod yet. Thanks a lot in advance! -- Léopold Baillard - Administrateur de Léoserveur - http://www.leoserveur.org/ Courriel : leobaill...@leoserveur.org Téléphone [FR] : + 33 (0) 6 20 32 16 32 Téléphone [DE] : + 49 (0) 151 21 40 55 46 GPG : 59C6 1CCA 2343 8DE4 D4FF D96A BC55 4A21 3B90 C658 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715
On 12/11/2012 10:12 PM, Mitul Limbani wrote: snom m9 dect ip But it's 2-3 x the price! 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
[asterisk-users] wcb4xxp extra hardware IDs
Hi Can anybody verify (confirm / deny) https://issues.asterisk.org/jira/browse/DAHLIN-220, which is actually a simple patch from http://bugs.debian.org/597760 ? It basically asks to add a few hardware IDs to the driver. But the report is a bit inconclusive. -- 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 -- 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 11 originate errors
I've observed the same behavior. This is what happens when you close the socket before logoff is completed. You need to wait until the logoff action is completed before closing the socket (your fclose() call). Alternately, use a proxy or a daemon that will sit between your script and the AMI and keep the socket open. On Wed, Dec 12, 2012 at 12:44 PM, Faheem faheem_...@yahoo.com wrote: Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1 OS: CentOS release 5.8 (Final) //manager.conf settings [faheem] secret =f@xx permit=127.0.0.1/255.255.255.255 read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan write = system,call,agent,user,config,command,reporting,originate ///AMI script ?php $sys_ip = 127.0.0.1; $User_str = faheem; $Secret_str = f@h33m112xx; $phoneNumb = 1234; $dialNumb = 4567; $spoofNumb = 786; $context = x-x; $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $User_str\r\n); fputs($oSocket, Secret: $Secret_str\r\n\r\n); fputs($oSocket, Events: off\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: SIP/testTrunk/$phoneNumb\r\n); fputs($oSocket, Exten: $dialNumb\r\n); fputs($oSocket, Context: $context\r\n); fputs($oSocket, Priority: 1\r\n\r\n); fputs($oSocket, Timeout: 1\r\n); fputs($oSocket, CallerId: $spoofNumb\r\n); fputs($oSocket, Async: false\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); echo originate executed; fclose($oSocket); ? Can any one please help me over it. Thank you! Muhammad Faheem -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Polycom phones and ring no answer/302 Moved Temporarily
I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for Always works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to No Answer means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a 302 SIP message issued. I can't find anything in the phone settings that look like it would disable this. Anyone else with a Polycom set that sees this, or does not see this and has forward no answer working? Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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] Polycom phones and ring no answer/302 Moved Temporarily
I have Polycom IP550. The Forward No Answer is working fine when enabled. I was looking at the sip.cfg but don't know exactly what to look for, can you give me a hint to where would i find that option? Thanks, On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for Always works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to No Answer means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a 302 SIP message issued. I can't find anything in the phone settings that look like it would disable this. Anyone else with a Polycom set that sees this, or does not see this and has forward no answer working? Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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] Need help designing implementation
Hi, I'd like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions. Take a look at gafachi (https://www.gafachi.com/), good voice quality and stable. Larry -- _ -- 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] Has iCall gone belly up? iCall carrier services bankrupt?
Hi everyone, Has iCall gone belly or just having really lazy executives / support team? They haven't placed a single long distance call for us since mid last month. Have they run away with deposit money? Are they bankrupt? I appreciate some feedback on this. Thanks, -Bruce -- _ -- 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
