Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashovabalas...@evaristesys.com wrote: Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard debug logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2_read: I should never be called - issue 8286
Hello ! I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi. The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri) and vice versa. After a period of time, I got the following scenario: NOTICE[860] chan_iax2.c: I should never be called! WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[860] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler NOTICE[860] chan_iax2.c: I should never be called! NOTICE[752] chan_iax2.c: I should never be called! Those messages has been spit out for a few seconds till asterisk crashed. I found bugreport #8286 with a similar symptom. It notes that the patch on this issues has been integrated. But I could not find anything similar in chan_iax2.c nor in channel.c in 1.6. Has this fix been pulled out again ? Thanks regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fastagi
Hello, I have a problem with fastagi. In fact I have a fastagi written in Java. Communcation between asterisk 1.6 and the server works correctly, except when a 'HANGUP' is sent by asterisk... In this case, the java server doesn't read the message. I have tride with PHP, same result. A ngrep show differences between the HANGUP messages and the others ( [AP] vs [AUP] ) T XXX.73.102.179:53824 - XXX.73.102.188:1687 [AP] 200 result=1 T XXX.73.102.179:53824 - XXX.73.102.188:1687 [AUP] HANGUP. Any idea? Kind regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fastagi
Appearances suggest that some part(s) of the AGI protocol changed between 1.4 and 1.6. hh174 wrote: Hello, I have a problem with fastagi. In fact I have a fastagi written in Java. Communcation between asterisk 1.6 and the server works correctly, except when a 'HANGUP' is sent by asterisk... In this case, the java server doesn't read the message. I have tride with PHP, same result. A ngrep show differences between the HANGUP messages and the others ( [AP] vs [AUP] ) T XXX.73.102.179:53824 - XXX.73.102.188:1687 [AP] 200 result=1 T XXX.73.102.179:53824 - XXX.73.102.188:1687 [AUP] HANGUP. Any idea? Kind regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Host-ID.
I'm about to change the motherboard in my server machine, (Different chipset). The most notable thing that will change, is the onboard network card (eth2) will be an atheros one instead of realtek. If I change the mac address of eth2 to read the same as the old one, will my host-id stay the same? TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on remote SIP calls
Hi The only time I've had issues that seem a bit like yours it was down to the order of codecs in the handset settings. Make sure they match the order dictated on the server. Ish Jonathan Moore wrote: On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien Cramattescrama...@zensoluciones.com wrote: Hi, This sounds udp RTP problem. Might be you have some firewall rules that block this kind of traffic ? As soon I remember, Asterisk by default use random port between 1 and 2 for rtp traffic (you can adjust this in rtp.conf). In theory, there should be no firewalls between my asterisk server and the remote phones. I've opened a ticket with ATT with that exact question, as well as a question of rather any NATing is going on, though, I doubt this is the case, and this is the first time this type of problem has happened in over 4 years. The idea of RTP being to blame would make sense though. I can still transfer and such, and watching the console, I see when I press various keys on the phone, so it seems that the SIP traffic is working out fine. (I do understand that right? SIP == control RTP == voice in a very generic sense?) I plan to take a packet trace in the morning on the asterisk server and see what is going on at that level. Hints as to what I should be looking for? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue agents get stuck
How do you do the log-on? l. 2009/8/6 Joao Gomes Pereira gomespere...@startel.pt Hello to all I have a queue where often my agents get stuck and cannot logoff. This is very bad, because agents cannot login again, and in Queuemetrics reports the agents appear to be online. How can I create a timeout to my agents and for the queue to kick them? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open source call center application for Asterisk
If you are completely new to Asterisk and want to run a professional call-center, my suggestion is to stick to a hand-made, lean, minimal configuration. l. 2009/7/13 ashish chauhan ashishchauhan07...@gmail.com Dear all, I am new to asterisk.i like to configure call center using asterisk.please can anyone tell me open source application to fulfill my requirement. thanks Ashish Kumar Chauhan M T S ,C D A C Chennai -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A problem with monitoring calls
Hello everyone, I have a problem with getting name of the recorded file of agent calls. As I've googled I found that the name of the recording file should be inserted in userfield of CDR table. To do this I set createlink=yes in agents.conf but still userfield of cdr is empty but the recrding file is created. my agents.conf file is like this(That part related to recording options): ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ; The format to be used to record the calls (wav, gsm, wav49) ; By default its wav. ;recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By default it's turned off. createlink=yes ; ; The text to be added to the name of the recording. Allows forming a url link. urlprefix=http://host.domain/calls/ ; ; The optional directory to save the conversations in. The default is ; /var/spool/asterisk/monitor ;savecallsin=/var/calls Has anybody any idea of what could be wrong? regards, Hooman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on remote SIP calls
Jonathan Moore supermegat...@gmail.com writes: The idea of RTP being to blame would make sense though. I can still transfer and such, and watching the console, I see when I press various keys on the phone, so it seems that the SIP traffic is working out fine. (I do understand that right? SIP == control RTP == voice in a very generic sense?) I plan to take a packet trace in the morning on the asterisk server and see what is going on at that level. Hints as to what I should be looking for? Start by looking at pure SIP traffic by doing -s0 -v and filtering on port 5060. Notice the media streams being negotiated, and look at the IP addresses and ports. If that doesn't help, remove the port 5060 filter and look again at the raw traffic -- but that can be a lot of traffic. My guess: You have STUN enabled on the phones. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available.. i tried it with Skype and it's great.. Asterisk and its great Callcentric VoIP provider and it was great.. one thing though i noticed that at some times you will have to dial again for the call to get setup. regards -- AHD Tarek Sawah Date: Thu, 6 Aug 2009 22:59:40 -0700 From: spamsucks2...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform? On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashov wrote: Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard debug logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get back to school stuff for them and cashback for you. http://www.bing.com/cashback?form=MSHYCBpubl=WLHMTAGcrea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
Fring works perfectly for me. Tarek Sawah a crit: Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available.. i tried it with Skype and it's great.. Asterisk and its great Callcentric VoIP provider and it was great.. one thing though i noticed that at some times you will have to dial again for the call to get setup. regards -- AHD Tarek Sawah Date: Thu, 6 Aug 2009 22:59:40 -0700 From: spamsucks2...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform? On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashov wrote: Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard "debug" logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get back to school stuff for them and cashback for you. http://www.bing.com/cashback?form=MSHYCBpubl=WLHMTAGcrea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crashes!!!
Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:40] WARNING[9517]: file.c:718 ast_readaudio_callback: Failed to write frame asterisk1*CLI Disconnected from Asterisk server ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
Guillermo Garron Alke Technology T. +591 33 141000 e. ggar...@alketech.com On 07/08/2009, at 06:41, hh174 oliv...@hh174.be wrote: Fring works perfectly for me. Tarek Sawah a écrit : Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available.. i tried it with Skype and it's great.. Asterisk and its great Callcentric VoIP provider and it was great.. one thing though i noticed that at some times you will have to dial again for the call to get setup. regards -- AHD Tarek Sawah you may also want to try nimbuzz.com also works with iPhone Date: Thu, 6 Aug 2009 22:59:40 -0700 From: spamsucks2...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform? On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashov wrote: Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard debug logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get back to school stuff for them and cashback for you. http://www.bing.com/cashback?form=MSHYCBpubl=WLHMTAGcrea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA922
Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA922
Show us your CLI output. I suspect that you're not getting a bridge and/or you're timing out. Also sip.conf and user.conf would be helpful as well as Asterisk release. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of robert boardman Sent: Friday, August 07, 2009 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linksys SPA922 Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host-ID.
AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Friday, August 07, 2009 3:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Host-ID. I'm about to change the motherboard in my server machine, (Different chipset). The most notable thing that will change, is the onboard network card (eth2) will be an atheros one instead of realtek. If I change the mac address of eth2 to read the same as the old one, will my host-id stay the same? TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhoneplatform?
I'm using it rather successfully. Not perfect, but it works. It is limited to WiFi connectivity... at least here in Spain I cant get either client to work over 3G. I'm using Fring and Truphone. Although I have only configured a SIP to my Asterisk with Fring. Skype works fine. We tested with several Nokia 5800 (EM) using Fring. Call quality is worse. At best, we have a 1+ second delay. Un saludo Enrique Mora Context M.I.S. SL em...@context.es -Mensaje original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de randulo Enviado el: viernes, 07 de agosto de 2009 7:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Anyone had any luck with SIP clients on the iPhoneplatform? Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good providers and one has unrecognizable audio and the other has noises as if the cable was badly soldered. I've never experienced such troubles with regular SIP clients. Anyone have any recommendations? Thanks /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4313 (20090806) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4313 (20090806) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help calls being dropped - maximum retries exceeded on trasmission
Hello all. I'm rather new. I'm lost and I would really appreciate if someone can point me in the right direction. I don't know if it something we did wrong or if it's an issue with our SIP TSP. We have a new Asterisk 1.4.26 server that has been running without a hitch for several days. I have two TSPs connected via SIP and one Linksys SPA-400 with 4 FXO ports. Calls are arriving correctly via the SPA-400 and one of our carriers. However, all calls from the other SIP operator are being dropped. Going through /var/log/messages I see that everything was working fine until suddenly, in the middle of the night, we started getting the following error messages on every call that arrived on one SIP trunk. [Aug 7 00:03:23] WARNING[7310] chan_sip.c: Maximum retries exceeded on transmission 17BC7D93-81FB11DE- b87df06a-ee73e...@xxx.yyy.119.39 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Aug 7 00:03:23] WARNING[7310] chan_sip.c: Hanging up call 17bc7d93-81fb11de-b87df06a-ee73e...@xxx.yyy.119.39 - no reply to our critical packet (see doc/sip-retransmit.txt). now I just get this (on every call from the faulty trunk) [Aug 7 17:00:24] WARNING[8308] chan_sip.c: Maximum retries exceeded on transmission 3f4276ab-829a11de-b7f6898e-f04...@xxx.22.119.35 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Aug 7 17:00:24] WARNING[8308] chan_sip.c: Hanging up call 3f4276ab-829a11de-b7f6898e-f04...@xxx.22.119.35 - no reply to our critical packet (see doc/sip-retransmit.txt). I did make a trivial change to the extensions.conf yesterday at 6pm but the system was taking calls correctly until midnight. All help is greatly appreciated. Regards, Enrique Enrique Mora em...@context.es ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host-ID.
Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Friday, August 07, 2009 3:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Host-ID. I'm about to change the motherboard in my server machine, (Different chipset). The most notable thing that will change, is the onboard network card (eth2) will be an atheros one instead of realtek. If I change the mac address of eth2 to read the same as the old one, will my host-id stay the same? TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
randulo a écrit : Hi, Hello I've tried two SIP clients so far and both have unusable outgoing audio quality. [...] Anyone have any recommendations? I made few test with various client, Sip and IAX, on iPhone first generation: . frings: good quality but to much delay. Also I don't like the fact that it's Frings server which register to Asteris, not the client. Question of privacy . iSip: good quality but also delay . siax: same as above, even better quality than iSip, but still delay, doesn't matter Sip or IAX . Weephone: perfect, good sound no delay. All those tests where made from one location to the same Asterisk server somewhere on Internet. Also I didn't pay attention on the look or if you can connect few accounts. Anyway, no one of those clients have the quality of a Nokia SIP client. To notice: when always WifI connected, the iPhone start to be hot, not cool when you're always on the phone ;-) -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] regcontext regexten
Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA922
Hi the asterisk version is 1.4.21.2 Here is the CLI -- Executing [...@incomming:1] Set(Zap/4-1, DB(lastcaller/zap4)=01942876818) in new stack -- Executing [...@incomming:2] GotoIf(Zap/4-1, 0?s-spoof|1:) in new stack -- Executing [...@incomming:3] Ringing(Zap/4-1, ) in new stack -- Executing [...@incomming:4] Set(Zap/4-1, CDR(accountcode)=s) in new stack -- Executing [...@incomming:5] Dial(Zap/4-1, SIP/105|20|tT) in new stack -- Called 105 Sip.conf ( with somethings changed) [gerneral] externhost=a.host.to.setup.com localnet=10.1.1.0/255.255.255.0 nat=yes [105] callerid=105 type=friend username=105 host=dynamic context=dialednum secret=red dtmfmode=rfc2833 disallow=all allow=alaw insecure=very ;mailbox=...@homr qualify=no nat=yes 2009/8/7 Danny Nicholas da...@debsinc.com Show us your CLI output. I suspect that you’re not getting a bridge and/or you’re timing out. Also sip.conf and user.conf would be helpful as well as Asterisk release. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *robert boardman *Sent:* Friday, August 07, 2009 9:01 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Linksys SPA922 Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -Mensaje original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de randulo Enviado el: viernes, 07 de agosto de 2009 17:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform? Ok, so now let me ask the question more directly: I am looking for the best SIP application for the iPod Touch (Wifi only). I don't care about 3g, Gsm or anything phone-related. The app has to be able to register with an arbitrary SIP service and/or dial arbitrary SIP URI. If it could dial one like this: 7463#2262...@proxy.ideasip.com That would be the ultimate app for me. If not, I can usually set up an alias on OnSIP (but they don't do # in the URI) I tried Fring, but it didn't seem optimal. I'll try again since many of you have said it works well. The two apps I mentioned both work and sound fine incoming, but the outgoing audio is either noisy (iSip) or inaudibly distorted. Note these apps are NOT free. ALso, I haven't tried contacting support yet, but I will do so after the conference call today. Thanks, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4315 (20090807) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4315 (20090807) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhoneplatform?
Ok, so now let me ask the question more directly: I am looking for the best SIP application for the iPod Touch (Wifi only). I don't care about 3g, Gsm or anything phone-related. The app has to be able to register with an arbitrary SIP service and/or dial arbitrary SIP URI. If it could dial one like this: 7463#2262...@proxy.ideasip.com That would be the ultimate app for me. If not, I can usually set up an alias on OnSIP (but they don't do # in the URI) I tried Fring, but it didn't seem optimal. I'll try again since many of you have said it works well. The two apps I mentioned both work and sound fine incoming, but the outgoing audio is either noisy (iSip) or inaudibly distorted. Note these apps are NOT free. ALso, I haven't tried contacting support yet, but I will do so after the conference call today. Thanks, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Going to VM after 180 seconds in queue
Hello all, This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info has [not] told me, the AGI doesn't allow for a timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? [1112221234] exten = s,1,Ringing exten = s,2,Wait(1) exten = s,3,Answer exten = s,4,GotoIfTime(09:00-21:00,mon-fri,*,*?s,8) exten = s,5,Background(closed) exten = s,6,Voicemail(2...@default) exten = s,7,Hangup() exten = s,8,Background(open) exten = s,9,AGI(agi-VDAD_ALL_inbound.agi,CID-LB-wholesale-1800234-Closer3-park--999-1) exten = _2100,1,Answer() exten = _2100,n,AGI(agi-VDAD_ALL_inbound.agi,CID-LB-2100-2100-Closer3-park--999-1) exten = _2100,n,Hangup() exten = _2101,1,Answer() exten = _2101,n,AGI(agi-VDAD_ALL_inbound.agi,CID-LB-2101-2101-Closer3-park--999-1) exten = _2101,n,Hangup() exten = _2102,1,Answer() exten = _2102,n,AGI(agi-VDAD_ALL_inbound.agi,CID-LB-2102-2102-Closer3-park--999-1) exten = _2102,n,Hangup() exten = i,1,Goto(s,4) exten = t,1,Goto(s,4) -Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
It depends on processor capability, disk access time and bandwidth. You will need to dedicate slices of disk and bandwidth for each machine. A realworld scenario of worst case would be this: You get sucky throughput on VM2 because 3 or 4 folks are monitoring calls or using voicemail on VM1. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Friday, August 07, 2009 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in VMWare,how does it perform and what is the limit? Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
been testing with Sun VirtualBox and i managed more than 30 extensions on a 2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring or encoding .. things went well -- AHD Tarek Sawah Date: Fri, 7 Aug 2009 08:47:03 -0700 From: jlama...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit? Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
Where you able to compile DAHDI in a virtual environment? How about skype for asterisk? Has anyone tried that in a virtual environment? Seems like to register the license, digium tool is looking for a connection on eth0, and in a virtual environment I see the name as vnet0 or vnet1. At least that what I see on godaddy's virtual servers. On Fri, Aug 7, 2009 at 12:08 PM, Tarek Sawah tareksa...@hotmail.com wrote: been testing with Sun VirtualBox and i managed more than 30 extensions on a 2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring or encoding .. things went well -- AHD Tarek Sawah Date: Fri, 7 Aug 2009 08:47:03 -0700 From: jlama...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit? Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host-ID.
On Friday 07 August 2009 10:11:23 Thomas Kenyon wrote: Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. Yes, it's based on all of them, and they should always present in the same order. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
Talk to damin AT nacs.net (he's on this mailinglist) Zoaaa James Lamanna wrote: Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host-ID.
Editing my original comment, linux uname should have been linux hostname. Tilghman, can you elaborate a bit more? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, August 07, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Host-ID. On Friday 07 August 2009 10:11:23 Thomas Kenyon wrote: Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. Yes, it's based on all of them, and they should always present in the same order. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com wrote: Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. So VMWare messes around with clock timing. This is a Bad Thing if you're trying to do things that rely on faithful timing, such as audio mixing for a MeetMe conference room. If you're only doing very simple things like playing messages or ordinary bridged two-way phone calls it probably wouldn't be as bad. If call quality matters, at all, I wouldn't go that route. If managing a real server with asterisk is too hard for your data center, may I humbly suggest an asterisk appliance? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ¡Xavier Cardil Coll te ha dejado un mensaje en Badoo!
¡Tienes un mensaje nuevo en Badoo! Xavier Cardil Coll te dejó un mensaje. Sigue el link para abrirlo: http://eu1.badoo.com/0153696585/in/ptA2fUgm9zo/?lang_id=7 Además, alguien ha estado preguntando por ti: Omar Lloper (Valencia, España)Laura Martin (Barcelona, España)Alberto Weingartshofer (Asunción, Paraguay) Si los links de este mensaje no funcionan, cópialos y pégalos en la barra de tu navegador. Este es un mensaje automáticamente generado. Las respuestas a este mensaje no serán leídas o respondidas. Si no deseas recibir más e-mails nuestros, haz click aquí: http://eu1.badoo.com/impersonation.phtml?lang_id=7mail_code=21email=asterisk-users%40lists.digium.comsecret=invite_id=175402user_id=153696585___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com wrote: Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. So VMWare messes around with clock timing. This is a Bad Thing if you're trying to do things that rely on faithful timing, such as audio mixing for a MeetMe conference room. If you're only doing very simple things like playing messages or ordinary bridged two-way phone calls it probably wouldn't be as bad. If call quality matters, at all, I wouldn't go that route. If managing a real server with asterisk is too hard for your data center, may I humbly suggest an asterisk appliance? Managing a server isn't the problem, I'm just looking to explore all solutions. If the call quality issues are that bad on vmware, then it is a non-starter in my book, especially trying to support the number of extensions I have now (I have 500 at the moment). Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Going to VM after 180 seconds in queue
On Friday 07 August 2009 11:04:14 Dan Pilcheck wrote: This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info has [not] told me, the AGI doesn't allow for a timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? You're not going to be able to do this without integration with the AGI. You could set an absolute timeout (TIMEOUT(absolute)), but that would fire regardless of whether the call was still in-queue or answered. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller id problem
I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Going to VM after 180 seconds in queue
Could you use AMI from within the AGI to poll the call status and act accordingly? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, August 07, 2009 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Going to VM after 180 seconds in queue On Friday 07 August 2009 11:04:14 Dan Pilcheck wrote: This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info has [not] told me, the AGI doesn't allow for a timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? You're not going to be able to do this without integration with the AGI. You could set an absolute timeout (TIMEOUT(absolute)), but that would fire regardless of whether the call was still in-queue or answered. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
On Fri, 2009-08-07 at 17:18 +0200, harry R wrote: Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 When this phone registers, Asterisk will automatically create an extension that looks like: exten = 6123,1,NoOp() in the [some-context] context. I use this in combination with DUNDi by setting the regcontext setting to point at my DUNDi advertising context, so that when my phone registers to a particular Asterisk server in my DUNDi cloud, calls get routed to the proper server. I'm sure there are other uses for it as well. For example, you might have something like this: exten = _6XXX,1,Playback(this-phone-is-not-registered) exten = 6123,2,Dial(SIP/myphone,20) exten = 6123,3,Voicemail(6...@default,u) Notice my priority numbering on extension 6123? If the phone is registered, then Asterisk creates priority number one for me. Otherwise, the pattern match plays a message saying that the phone is not registered. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
Yes, the issue(s) is/are: 1. The VOIP provider may be masking the callerID for their own cost allocation reasons. That is some of the issue. 2. Your Asterisk box may forward some of the regular phone line calls with their caller ID. 3. Somehow, the number you want to use may leak through sometimes. :-) What you need to do is put in a simple, absolute CallerID(num) = 3216540987 type of statement before sending the call out. Make it apply to every call no matter what. That isn't the syntax but you get the idea. Of course you won't have true caller ID then, but do you want cheap or real? Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan Sent: Friday, August 07, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] caller id problem I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Going to VM after 180 seconds in queue
On Fri, 7 Aug 2009, Dan Pilcheck wrote: This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info has [not] told me, the AGI doesn't allow for a timeout to be set. Would setting an absolute timeout at the start of the AGI and clearing it when the call is answered allow you to use the T extension to catch the timer expiring for the non-answered calls? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
Hi Cary, Thanks for the quick reply :D I get what you're saying. I have a suspicion that it is the telco's fault since every other number that receives a call from my Asterisk box displays the correct number. I'll give setting the caller id another go and play with that. I guess what I am looking for is a) confirmation that this problem has happened to other people and b) a suggestion of how to point the tech support in the right direction so they can fix this problem for me, or how I can just override this problem myself. Thanks again for your help and quick reply. Cary Fitch wrote: Yes, the issue(s) is/are: 1. The VOIP provider may be masking the callerID for their own cost allocation reasons. That is some of the issue. 2. Your Asterisk box may forward some of the regular phone line calls with their caller ID. 3. Somehow, the number you want to use may leak through sometimes. :-) What you need to do is put in a simple, absolute CallerID(num) = 3216540987 type of statement before sending the call out. Make it apply to every call no matter what. That isn't the syntax but you get the idea. Of course you won't have true caller ID then, but do you want cheap or real? Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan Sent: Friday, August 07, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] caller id problem I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
So far, the best iPhone platform app I've found is a $10 one called iPico. It is a one account SIP client, better designed than the others and it actually works and can dial SIP URI. I learned about it directly from Ruben Olsen mentioning it on the VUC call an hour ago. I will be posting the edited recording of the session later today on http://VUC.me but it is available on Talkshoe now: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 We had several technical issues that I will edit out later today, but there it is, warts and all. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime config and extensions.conf
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include = trunktollfree
[longdistance]
include = local
include = trunkld
[international]
include = longdistance
include = trunkint
[from-pstn]
exten = 7157999,1,VoicemailMain()
switch = Realtime
[residential]
include = from-pstn
include = international
---
And the relevant entries in the DB:
mysql select name, context from sip_buddies;
+-+-+
| name| context |
+-+-+
| 7157986 | residential |
| 7157980 | residential |
+-+-+
2 rows in set (0.01 sec)
mysql select * from extensions;
++-+-+--++-+
| id | context | exten | priority | app| appdata |
++-+-+--++-+
| 10 | residential | 7157986 |1 | Dial | SIP/7157986 |
| 11 | residential | 7157986 |2 | Congestion | |
| 12 | residential | 7157980 |1 | Dial | SIP/7157980 |
| 13 | residential | 7157980 |2 | Congestion | |
++-+-+--++-+
4 rows in set (0.00 sec)
---
The phone I am testing with has a sip entry in sip_buddies with a
context of residential. As you can see from the cascading contexts
above the residential context can dial local 7 digit numbers via the
TRUNK (a zap T1 with an inbound context of from-pstn), but dialing the
Voicemail main number, also seven digits, overrides this and is executed
directly. This all works as expected and seems fairly elegant.
I also expected that the switch = Realtime statement in [from-pstn]
would allow any local numbers in the extensions table to also override
the trunk dialing, but it does not. So my test phone, when it dials a
local number that exists in the extensions table, ends up sending the
call out the TRUNK, then it comes back in the TRUNK on another channel,
and then dials the SIP phone as expected. The call at least goes through
:) But it does kill the video H.264 stream I was hoping for!
How can I make sure that the realtime entries override the pattern
matching in [trunk-local]?
Thanks,
j
___
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Re: [asterisk-users] realtime config and extensions.conf
Meant to add that this is 1.4.26... :)
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include = trunktollfree
[longdistance]
include = local
include = trunkld
[international]
include = longdistance
include = trunkint
[from-pstn]
exten = 7157999,1,VoicemailMain()
switch = Realtime
[residential]
include = from-pstn
include = international
---
And the relevant entries in the DB:
mysql select name, context from sip_buddies;
+-+-+
| name| context |
+-+-+
| 7157986 | residential |
| 7157980 | residential |
+-+-+
2 rows in set (0.01 sec)
mysql select * from extensions;
++-+-+--++-+
| id | context | exten | priority | app| appdata |
++-+-+--++-+
| 10 | residential | 7157986 |1 | Dial | SIP/7157986 |
| 11 | residential | 7157986 |2 | Congestion | |
| 12 | residential | 7157980 |1 | Dial | SIP/7157980 |
| 13 | residential | 7157980 |2 | Congestion | |
++-+-+--++-+
4 rows in set (0.00 sec)
---
The phone I am testing with has a sip entry in sip_buddies with a context
of residential. As you can see from the cascading contexts above the
residential context can dial local 7 digit numbers via the TRUNK (a zap T1
with an inbound context of from-pstn), but dialing the Voicemail main
number, also seven digits, overrides this and is executed directly. This all
works as expected and seems fairly elegant.
I also expected that the switch = Realtime statement in [from-pstn]
would allow any local numbers in the extensions table to also override the
trunk dialing, but it does not. So my test phone, when it dials a local
number that exists in the extensions table, ends up sending the call out
the TRUNK, then it comes back in the TRUNK on another channel, and then dials
the SIP phone as expected. The call at least goes through :) But it does
kill the video H.264 stream I was hoping for!
How can I make sure that the realtime entries override the pattern matching
in [trunk-local]?
Thanks,
j
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
I was able to get a VMWare Fusion CentOS 5.3 with Asterisk 1.6.0.9 talking to a Xorcom Astribank on my MacBook. I could connect a POTS line to an FXO port and a phone to an FXS port and make calls. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Aug 7, 2009, at 9:25 AM, Pascal Bruno wrote: Where you able to compile DAHDI in a virtual environment? How about skype for asterisk? Has anyone tried that in a virtual environment? Seems like to register the license, digium tool is looking for a connection on eth0, and in a virtual environment I see the name as vnet0 or vnet1. At least that what I see on godaddy's virtual servers. On Fri, Aug 7, 2009 at 12:08 PM, Tarek Sawah tareksa...@hotmail.com wrote: been testing with Sun VirtualBox and i managed more than 30 extensions on a 2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring or encoding .. things went well -- AHD Tarek Sawah Date: Fri, 7 Aug 2009 08:47:03 -0700 From: jlama...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit? Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime config and extensions.conf
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include = trunktollfree
[longdistance]
include = local
include = trunkld
[international]
include = longdistance
include = trunkint
[from-pstn]
exten = 7157999,1,VoicemailMain()
switch = Realtime
[residential]
include = from-pstn
include = international
---
And the relevant entries in the DB:
mysql select name, context from sip_buddies;
+-+-+
| name| context |
+-+-+
| 7157986 | residential |
| 7157980 | residential |
+-+-+
2 rows in set (0.01 sec)
mysql select * from extensions;
++-+-+--++-+
| id | context | exten | priority | app| appdata |
++-+-+--++-+
| 10 | residential | 7157986 |1 | Dial | SIP/7157986 |
| 11 | residential | 7157986 |2 | Congestion | |
| 12 | residential | 7157980 |1 | Dial | SIP/7157980 |
| 13 | residential | 7157980 |2 | Congestion | |
++-+-+--++-+
4 rows in set (0.00 sec)
---
The phone I am testing with has a sip entry in sip_buddies with a context
of residential. As you can see from the cascading contexts above the
residential context can dial local 7 digit numbers via the TRUNK (a zap T1
with an inbound context of from-pstn), but dialing the Voicemail main
number, also seven digits, overrides this and is executed directly. This all
works as expected and seems fairly elegant.
I also expected that the switch = Realtime statement in [from-pstn]
would allow any local numbers in the extensions table to also override the
trunk dialing, but it does not. So my test phone, when it dials a local
number that exists in the extensions table, ends up sending the call out
the TRUNK, then it comes back in the TRUNK on another channel, and then dials
the SIP phone as expected. The call at least goes through :) But it does
kill the video H.264 stream I was hoping for!
How can I make sure that the realtime entries override the pattern matching
in [trunk-local]?
Thanks,
j
And now to answer my own silly question...
The switch statement will use the static context it is a member of to
search the tables, and I had 'residential' rather than 'from-pstn' in the
tables.
Works fine now :)
Cheers,
j
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] Host-ID.
Danny Nicholas wrote: Editing my original comment, linux uname should have been linux hostname. Tilghman, can you elaborate a bit more? It's definitely not based on that either since changing your hostname doesn't change your Host-ID. In case anyone was wondering, I changed the adapter address on the new board so that it matched the old one and got udev to make sure it had the same name. Then started asterisk and my licenses were in tact. I didn't check what the host-Id was before doing this though. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, August 07, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Host-ID. On Friday 07 August 2009 10:11:23 Thomas Kenyon wrote: Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. Yes, it's based on all of them, and they should always present in the same order. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Placing a SIP Call on Hold
I want to a place a call (SIP) on hold in asterisk? Is there any way to do it? If yes, please give me an example. We are using Asterisk 1.4.24.1. Any help would be appreciated... Thanks Regards, Venkat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Placing a SIP Call on Hold
Does it this link help? http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold On Fri, Aug 7, 2009 at 10:07 PM, Venkateshwarlu Kakkirenivenka...@iconsultech.com wrote: I want to a place a call (SIP) on hold in asterisk? Is there any way to do it? If yes, please give me an example. We are using Asterisk 1.4.24.1. Any help would be appreciated... Thanks Regards, Venkat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
If you want to hang more results on this subject, please see the thread here: http://www.voipusersconference.org/2009/08/sip-for-apple-iphone/ I'm very interested in anyone who is doing development in this space so keep in touch. Basically, even though I've always preferred DECT/SIP phones to wifi/SIP ones, a good SIP client adds the wifi capability to an otherwise very good mp3 and video player, provided you invest in a new headset with a mic. So far, so good. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Placing a SIP Call on Hold
Thanks for a quick reply... This link just shows how to set MOH feature if the phone has hold feature. I want to place a call on hold irrespective of SIP phones used... If I create an MOH extension as shown transfer the calls to that extension and then if one party disconnects the call, the other party is still hearing the MOH... Thanks Regards, Venkat -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 08 August 2009 02:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Placing a SIP Call on Hold Does it this link help? http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold On Fri, Aug 7, 2009 at 10:07 PM, Venkateshwarlu Kakkirenivenka...@iconsultech.com wrote: I want to a place a call (SIP) on hold in asterisk? Is there any way to do it? If yes, please give me an example. We are using Asterisk 1.4.24.1. Any help would be appreciated... Thanks Regards, Venkat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca wrote: I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. Since this is already a little off-topic, care to share which provider you are using? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
David Backeberg wrote: On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca wrote: I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. Since this is already a little off-topic, care to share which provider you are using? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah, no problem. The telco is Telus in British Columbia, Canada and Digital Voice in Vancouver is my VOIP provider. I'd rather not have to switch telcos as there is always some nice fees and charges when you sign up. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Going to VM after 180 seconds in queue
2009/8/8 Dan Pilcheck pilch...@gmail.com Hello all, This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info has [not] told me, the AGI doesn't allow for a timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? How about just setting the hold time in vicidial to 3 minutes, specifying a drop action of extension and specifying drop extension of 2100? d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 30 Great free Asterisk applications
Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few programs in the list that even the most seasoned Asterisk professionals haven't seen before: http://www.venturevoip.com/news.php?rssid=2184 -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
