[asterisk-users] sip calls not going through
Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call goes through and voice is perfect.. and at others one side can hear me yet i cant hear them.. and at others neither one of us can hear the other end.. i've checked my logs and havent found anything relevant.. but yet again maybe you could as i'm a newbie.. Registered SIP '101' at 192.168.75.192 port 22162 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 101 -- Executing [...@spa:1] Dial(SIP/100-0967ad88, SIP/101|15) in new stack -- Called 101 -- SIP/101-09683690 is ringing -- SIP/101-09683690 answered SIP/100-0967ad88 -- Native bridging SIP/100-0967ad88 and SIP/101-09683690 == Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-0967ad88' -- Executing [...@spa:1] Dial(SIP/101-09676378, SIP/100|15) in new stack -- Called 100 -- SIP/100-0967ad88 is ringing -- SIP/100-0967ad88 answered SIP/101-09676378 -- Native bridging SIP/101-09676378 and SIP/100-0967ad88 == Spawn extension (spa, 100, 1) exited non-zero on 'SIP/101-09676378' -- Executing [...@spa:1] Dial(SIP/100-09676378, SIP/101|15) in new stack -- Called 101 -- SIP/101-09677b10 is ringing -- SIP/101-09677b10 answered SIP/100-09676378 -- Native bridging SIP/100-09676378 and SIP/101-09677b10 == Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-09676378' -- Unregistered SIP '100' -- Registered SIP '100' at 192.168.75.139 port 14226 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 100 -- Executing [...@spa:1] Dial(SIP/100-096792c8, SIP/101|15) in new stack -- Called 101 -- SIP/101-09683690 is ringing -- SIP/101-09683690 answered SIP/100-096792c8 -- Native bridging SIP/100-096792c8 and SIP/101-09683690 == Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-096792c8' -- Unregistered SIP '100' -- Registered SIP '100' at 192.168.75.139 port 41372 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 100 -- Executing [...@spa:1] Dial(SIP/100-096792c8, SIP/101|15) in new stack -- Called 101 -- SIP/101-09683690 is ringing -- SIP/101-09683690 answered SIP/100-096792c8 -- Native bridging SIP/100-096792c8 and SIP/101-09683690 == Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-096792c8' -- Unregistered SIP '100' -- Executing [...@spa:1] Dial(SIP/101-09677b10, SIP/100|15) in new stack [Jun 10 09:37:54] WARNING[7880]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@spa:2] VoiceMail(SIP/101-09677b10, 1...@default) in new stack -- SIP/101-09677b10 Playing 'vm-intro' (language 'en') -- Registered SIP '100' at 192.168.75.139 port 58704 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 100 -- SIP/101-09677b10 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/100/tmp/zWOOql format: wav, 0x967d7b0 -- User hung up == Spawn extension (spa, 100, 2) exited non-zero on 'SIP/101-09677b10' -- Executing [...@spa:1] Dial(SIP/101-09677b10, SIP/100|15) in new stack -- Called 100 -- SIP/100-0967ad88 is ringing -- SIP/100-0967ad88 answered SIP/101-09677b10 -- Native bridging SIP/101-09677b10 and SIP/100-0967ad88 == Spawn extension (spa, 100, 1) exited non-zero on 'SIP/101-09677b10' -- Unregistered SIP '100' -- Registered SIP '100' at 192.168.75.139 port 14744 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 100 -- Unregistered SIP '101' -- Executing [...@spa:1] Dial(SIP/100-096792c8, SIP/101|15) in new stack [Jun 10 09:39:23] WARNING[7885]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@spa:2] VoiceMail(SIP/100-096792c8, 1...@default) in new stack -- SIP/100-096792c8 Playing 'vm-intro' (language 'en') == Spawn extension (spa, 101, 2) exited non-zero on 'SIP/100-096792c8' -- Registered SIP '101' at 192.168.75.192 port 34518 -- Saved useragent X-Lite release 1014k stamp 47051 for peer 101 -- Executing [...@spa:1] Dial(SIP/100-0967f670, SIP/101|15) in new stack -- Called 101 -- SIP/101-09684cb0 is ringing -- SIP/101-09684cb0 answered SIP/100-0967f670 -- Native bridging SIP/100-0967f670 and SIP/101-09684cb0 == Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-0967f670' -- Executing [...@spa:1] Dial(SIP/101-0967f670, SIP/100|15) in new stack -- Called
[asterisk-users] Advice
Hi all, a few month ago I got the task of setting up asterisk for my company. I had 94 employee to set this up for ... I never heard of asterisk before to b honest, so after researching a bit.. I started with a digium card with ZAP though that didn't work out as the card were flawed.. so ended up setting up SIP for everyone using a SIP callcentric accounts as well as sipura for pstn lines.. now it's working at it's minimal state.. but as am out of the heat of pressure from management.. so now It's time to learn about asterisk the right way as I had lots of help from this mailing list as well as the IRC channel that I'm not sure I could do it again on my own.. so not to add more to my email, I'm seeking advice about the proper way to learn about asterisk from A to Z if possible... any advice would be appreciated thanks in advance, RolandEmoticon1.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
Hi Michel, how's beirut's weather with ya! anyway, TTL stands for TIME To LIVE. it's encapsulated on layer three of the OSI layer to each packet going out that specific interface. by default routers has a 16 TTL that means each time the designated packet reaches a router (gets decapsulated) it gets a -1... this helps in preventing loops which would eventually lead to congestion. now latency wise, for VOIP to operate correctly it needs a latency of under 200 ms. (I currently have a microwave link , and unfortunately im not getting that a latency less than 280 to my SIP provider) if your asterisk server is hosted online, you could simply traceroute it and check the highest latency, point. and depending on where that bottle neck would be, youll troubleshoot from there.. mine were on my ISP's international link, after having a meeting with my account manager, I got my link routed through a different international path which drastically decreased my latency. now on a different approach, you absolutly have to talk to your ISP/network administrator to provide you QOS for that specific IP whether it's public or private. depending on your network's traffic QOS would surely help with no doubt.. this would decrease latency as well hope I've shed some light about this, if not well the more knowledge the betteR best, Roland From: michel freiha Sent: Monday, December 15, 2008 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users-boun...@lists.digium.com Subject: Re: [asterisk-users] tcpdum Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip to sip unplanned conference! help!!
first of all my topology is as such:Softphones-- asterisk -- sipurasoftphone with peer number 100, calls another softphone with peer number as 200. (both has asterisk as gateway)relevant extensions.conf: exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line is busy or unavailable exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line is busy or unavailable exten = _2XX,3,HangUp() relevant sip.conf: [200] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [200] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] in the meantime, an incoming call comes through Sipura which is directed to: [incoming-samer] exten = 201,1,Answer() ; Answer inbound calls exten = 201,2,Playback(silence/1) exten = 201,3,Background(joyce) ; input an extension exten = 201,4,WaitExten(8) exten = 201,5,Dial(SIP/220,15) exten = 201,4,Wait(8) include = spa exten = 201,n,Hangup() suddenly, the first conversation between 100 and 200, hears the attendant audio message joyce welcoming the caller(the one calling sipura in a completely different call) and listens to the entire conversation that the incoming caller is having.. _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] sip conversations overlapping!!!!
i appologize for not making myself clear.. i have my asterisk box, connexted to 4 sipura3102.. these sipuras has 4 PSTN lines connected to them through one cable, which has 8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs port in the sipura) on the other side, i have 20 softphones.. these softphones has asterisk as their gateway.. where they could call eachother! or call/recieve calls through any of the sipuras... my prob is as such: when i call from softphone#1 to sipura #1, sound is pretty good and everything is working perfectly.. though if asterisk recieves a call from another sipura.. lets say its sipura #2, then! i could hear the attendnat answering the incoming phone in my current conversation, and i could hear some1 picking up and answerinfg the call..! if i ask them to hang up! my line breaks as well.. Date: Fri, 29 Aug 2008 10:40:57 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip conversations overlapping Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! Not clear for me, develop some more you topology. _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] sip conversations overlapping!!!!
Hi all, i'm facing this weird prob...my topology is as such: softphone --- asterisk sipura 3102 sipura 3102 -sipura 3102 -sipura 3102 when am on a call, sometimes when some1 else tries to call out.. i hear the actual tones which ends up preventing the other end from talking to me.. moroever, when some1 calls me through one sipura, while im talking on another... i can hear the attendant welcoming message, then i hear the voice of whoever have picked tht line up..! and if i ask that person to hang up... my line breaks as well..! can any1 help me with this issue! below is my config: extensions.conf [incoming-conference] exten = 333,1,Answer() ; Answer inbound calls exten = 333,2,Playback(silence/1) exten = 333,3,Background(joyce) ; input an extension exten = 333,4,WaitExten(8) exten = 333,5,Dial(SIP/310,15) exten = 333,4,Wait(8) include = spa exten = 333,n,Hangup() [incoming-samer] exten = 334,1,Answer() ; Answer inbound calls exten = 334,2,Playback(silence/1) exten = 334,3,Background(joyce) ; input an extension exten = 334,4,WaitExten(8) exten = 334,5,Dial(SIP/330,15) exten = 334,4,Wait(8) include = spa exten = 334,n,Hangup() [incoming-gilberte] exten = 335,1,Answer() ; Answer inbound calls exten = 335,2,Playback(silence/1) exten = 335,3,Background(joyce) ; input an extension exten = 335,4,WaitExten(8) exten = 335,5,Dial(SIP/350,15) exten = 335,4,Wait(8) include = spa exten = 335,n,Hangup() [incoming-line4] exten = 336,1,Answer() ; Answer inbound calls exten = 336,2,Playback(silence/1) exten = 336,3,Background(joyce) ; input an extension exten = 336,4,WaitExten(8) exten = 336,5,Dial(SIP/340,15) exten = 336,4,Wait(8) include = spa exten = 336,n,Hangup() [sipura-line] exten = 301,1,Answer() ; Answer inbound calls exten = 301,2,Playback(silence/1) exten = 301,3,Background(simzy1) ; input an extension exten = 301,4,WaitExten(8) exten = 301,5,Dial(SIP/100,15) ; goes to operator exten = 301,4,Wait(8) include = spa [spa] exten =_301,1,GoTo(sipura-line,${EXTEN},1) exten =_333,1,GoTo(incoming-conference,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line is busy or unavailable exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line is busy or unavailable exten = _2XX,3,HangUp() exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it will ring 3 times exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line is busy or unavailable exten = _3XX,3,HangUp() exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line ;exten =_01,2,Set(TIMEOUT(absolute)=5) exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4 exten = 303,1,VoicemailMain ; voicemail box to be redirected to sip.conf: [300] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] canreinvite=yes [301] type=friend host=dynamic secret=1234 context=sipura-line [EMAIL PROTECTED] [304] type=friend host=dynamic secret=1234 context=sipura-line2 [EMAIL PROTECTED] [305] type=friend host=dynamic secret=1234 context=incoming-samer [EMAIL PROTECTED] [306] type=friend host=dynamic secret=1234 context=incoming-gilberte [EMAIL PROTECTED] [333] type=friend host=dynamic secret=1234 context=incoming-conference [EMAIL PROTECTED] [334] type=friend host=dynamic secret=1234 context=incoming-samer [EMAIL PROTECTED] [335] type=friend host=dynamic secret=1234 context=incoming-gilberte [EMAIL PROTECTED] [336] type=friend host=dynamic secret=1234 context=incoming-line4 [EMAIL PROTECTED] [307] type=friend host=dynamic secret=1234 context=incoming-conference [EMAIL PROTECTED] [308] type=friend host=dynamic secret=1234 context=incoming-line4 [EMAIL PROTECTED] [310] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [320] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [330] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [340] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [350] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [107] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [150] type=friend host=dynamic secret=1234 context=spa [EMAIL PROTECTED] [100] type=friend host=dynamic
Re: [asterisk-users] sip conversations overlapping!!!!
Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! Date: Thu, 28 Aug 2008 14:10:53 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip conversations overlapping RoLaNd RoLaNd wrote: Hi all, i'm facing this weird prob...my topology is as such: softphone --- asterisk sipura 3102 sipura 3102 -sipura 3102 -sipura 3102 when am on a call, sometimes when some1 else tries to call out.. i hear the actual tones which ends up preventing the other end from talking to me.. moroever, when some1 calls me through one sipura, while im talking on another... i can hear the attendant welcoming message, then i hear the voice of whoever have picked tht line up..! and if i ask that person to hang up... my line breaks as well..! can any1 help me with this issue! below is my config: How are the analogue phones wired? One phone plugged directly to one 3102 FXS port? or is there common wiring ? Are all the FXO ports connected to telco lines? regards, Drew NOTE: Holding the SHIFT key down whilst typing the first person, singular, pronoun will produce stunningly readable results. Either SHIFT key will do, you can even use the CAPS LOCK key if both of those are broken/can't locate them. You can also use this procedure for the first letter of each sentence, it makes everything much easier to read. -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
i kinda have a relevant prob! my sipura 3102 wont pass CID to asterisk even though ive enabled such a feature in its web gui! Date: Wed, 27 Aug 2008 12:07:51 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura Does anybody have an idea how to pass Off Hook caller ID to Asterisk via Linksys ? I'm getting caller ID type I OK but when another customer rings the phone (when I'm on line) the CID off hook is not coming through. I think Off-Hook CID is called CID type II, isn't it? -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] entering a password to have access to a sip account?!
Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right now if i dial 03 it rings at the other end! though i dont want that to happen! i want to set asterisk up in a way tht if i dial 03 from 300 to ask me for a password... or it wont let the line go through! can anyone guide me through this issue! im really going crazy to get this done! any help would truly and utterly be appreciated:) ps: find below my extensions.conf [sipura-line] exten = 301,1,Answer() ; Answer inbound calls exten = 301,2,Playback(silence/1) exten = 301,3,Background(simzy1) ; input an extension exten = 301,4,WaitExten(8) exten = 301,5,Dial(SIP/100,15) ; goes to operator exten = 301,4,Wait(8) include = spa exten = _XXX,6,VoiceMail([EMAIL PROTECTED]) exten = 301,n,Hangup() [spa] exten =_301,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line is busy or unavailable exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line is busy or unavailable exten = _2XX,3,HangUp() exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it will ring 3 times exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line is busy or unavailable exten = _3XX,3,HangUp() exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line ;exten =_01,2,Set(TIMEOUT(absolute)=5) exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4 exten = 303,1,VoicemailMain ; voicemail box to be redirected to _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
Hello Steve, thanks for the advice :) though one prob! if i add the authenticate line itll require all callers to enter 1234 to access *ANY* sip account.. even though this would come in handy at some point but at the moment i just want to deny the extension 300 from being able to call 01 unless the caller entered a password.. find below wht i did so far.. [sipura-line] exten = 301,1,Answer() ; Answer inbound calls exten = 301,2,Playback(silence/1) exten = 301,3,Background(simzy1) ; input an extension exten = 301,4,authenticate(1234) exten = 301,5,WaitExten(8) exten = 301,6,Dial(SIP/100,15) ; goes to operator exten = 301,3,Wait(8) include = spa exten = _XXX,6,VoiceMail([EMAIL PROTECTED]) exten = 301,n,Hangup() [spa] exten =_301,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line is busy or unavailable exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line is busy or unavailable exten = _2XX,3,HangUp() exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it will ring 3 times exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line is busy or unavailable exten = _3XX,3,HangUp() exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line ;exten =_01,2,Set(TIMEOUT(absolute)=5) exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer exten = 303,1,VoicemailMain ; voicemail box to be redirected to Date: Sun, 24 Aug 2008 12:05:02 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] entering a password to have access to a sip account?! You want to use Authenticate() between answer and dial. http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a Thanks, Steve Totaro On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote: Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right now if i dial 03 it rings at the other end! though i dont want that to happen! i want to set asterisk up in a way tht if i dial 03 from 300 to ask me for a password... or it wont let the line go through! can anyone guide me through this issue! im really going crazy to get this done! any help would truly and utterly be appreciated:) ps: find below my extensions.conf [sipura-line] exten = 301,1,Answer() ; Answer inbound calls exten = 301,2,Playback(silence/1) exten = 301,3,Background(simzy1) ; input an extension exten = 301,4,WaitExten(8) exten = 301,5,Dial(SIP/100,15) ; goes to operator exten = 301,4,Wait(8) include = spa exten = _XXX,6,VoiceMail([EMAIL PROTECTED]) exten = 301,n,Hangup() [spa] exten =_301,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line is busy or unavailable exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line is busy or unavailable exten = _2XX,3,HangUp() exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it will ring 3 times exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line is busy or unavailable exten = _3XX,3,HangUp() exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line ;exten =_01,2,Set(TIMEOUT(absolute)=5) exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4 exten = 303,1,VoicemailMain ; voicemail box to be redirected to Get news, entertainment and everything you care about at Live.com. Check it out! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] 5 min limitation on phone calls! how to!
Hello all! my last month's phone bill sky rocketed after i setup asterisk with softphones all over the house! could someone help me set up a limitation for my wife and kids not to be able to talk for more than 5 min at a time! or like 20 min per week! or whtever limitation i could set for this! any help would trull be appreciated:) _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] opening Doors with Asterisk!?
Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] list of minutes spent on SIP phone calls?! any advice?!
Hi All, i have asterisk with 9 SIP accounts on it. i was wondering if theres a way to setup asterisk, to send the amount of minutes each SIP account have spent incoming as well as outgoing and if possible the number it called! any advice?! any help would truly be appreciated..! thanks in advance and best regards, _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] removing == Parsing '/etc/asterisk/manager.conf': Found from CLI!
hi all, is there any way of removing this line from showing on the console? my verbosity level is 3. and this is the following output on cli 24/7 unless its interrupted by the msgs tht really counts like connected sip and so on.. [Jul 4 10:32:38] NOTICE[18542]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:48] NOTICE[18543]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:58] NOTICE[18544]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:08] NOTICE[18545]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate -- Registered SIP '179' at 192.168.0.2 port 27780 expires 3600 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:18] NOTICE[18547]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] how to setup one stage dialing plan, instead of two! help!!!
Hello all, i recently finished setting up my asterisk with sipura 3102 using PSTN. this is my dial plan relevant to wht i want: exten =_01,1,Dial(SIP/$(EXTEN)@200) right now as u see i made my dial plan on a 2 stage dialing mode. tht means i dial 01, i get the pstn dial tone, and then i call whichever number i want through it. i want to have the option for my call to directly go through pstn without having to wait for the pstn dial tone. any help would be appreciated.. :) _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Windows Mobile 6 IAX/SIP client?
Hey! i'm facing the same prob.. i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip client..! so far i found these 3: AGEphone mobile: http://www.ageet.com/ SJphone: http://www.sjlabs.com/sjp.html Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html so far i just tried AgePhone (trial mode) sound is great though im facing 2 problems with it: 1: u can only use handsfree option, tht mean its a privacy killer. 2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your asterisk (if u call an extension, and wait to hear a dialtone) it wont work. as for the other 2 i didnt try them yet... ps: if you found out anything else bout this matter id appreciate if you could let me know :) Date: Mon, 30 Jun 2008 21:51:57 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client? I just bought a HTC TyTn II phone, but unfortunately it doesn't even have a SIP client in it. I tried the wiki searching for a SIP or IAX client but only found some PocketPC stuff (Windows Mobile 2003). Does anyone know of a good quality SIP or IAX softphone that will run on Windows Mobile 6? I only have a data subscription, no voice so the quality should be sufficient to be used constantly. Thanks!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Windows Mobile 6 IAX/SIP client?
Hey Matt!! thanks for the advice! appreciate it.. just installed it and everything worked fine ( i got internet calling in my menu) though i cant seem to access the editing tool.. keeps on giving me some error even after soft reseting.. any idea?! From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Thu, 3 Jul 2008 13:11:40 -0400 Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client? Hi Roland, Did you try: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/ We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II) Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd Sent: Thursday, July 03, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client? Hey! i'm facing the same prob.. i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip client..! so far i found these 3: AGEphone mobile: http://www.ageet.com/ SJphone: http://www.sjlabs.com/sjp.html Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html so far i just tried AgePhone (trial mode) sound is great though im facing 2 problems with it: 1: u can only use handsfree option, tht mean its a privacy killer. 2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your asterisk (if u call an extension, and wait to hear a dialtone) it wont work. as for the other 2 i didnt try them yet... ps: if you found out anything else bout this matter id appreciate if you could let me know :) Date: Mon, 30 Jun 2008 21:51:57 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client? I just bought a HTC TyTn II phone, but unfortunately it doesn't even have a SIP client in it. I tried the wiki searching for a SIP or IAX client but only found some PocketPC stuff (Windows Mobile 2003). Does anyone know of a good quality SIP or IAX softphone that will run on Windows Mobile 6? I only have a data subscription, no voice so the quality should be sufficient to be used constantly. Thanks!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 Discover the new Windows Vista Learn more! _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] incoming calls through callcentric sip account!!
Hi all, i've recently acquired a callcentric account. i've perfectly setup my sip.conf and extensions.conf to make outgoing calls. but the problem is with incoming calls! when i call in, asterisk doesnt even see the incoming call! how is tht possible! please see the following my config: sip.conf: [general] dtmfmode = rfc2833 context=from-callcentric srvlookup=yes register = username:[EMAIL PROTECTED]/username [callcentric] type=peer context=from-callcentric host=callcentric.com username=username secret=password fromuser=username fromdomain=callcentric.com insecure=very [107] context=to-callcentric type=friend username=107 secret=secret host=dynamic etensions.conf: [from-callcentric] exten = s,1,Dial(SIP/107) [to-callcentric] exten = _0.,1,Dial(SIP/[EMAIL PROTECTED]) _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] extensions help!
hello all, was wondering if some1 could help me to add an option to my incoming operator menu. currently, when some1 calls in, he gets a recorded msg asking for him to punch in an extension or dial 100 for operator assistance wht i want is to add 2 other things; firstly, if in a period of time the person didnt punch in an extension i want him to b directed atomaticly to the operator. 2ndly, to add an option of lets say, press 2 to listen to availabe extensions this is my extensions.conf [sipura-line] exten = 201,1,Answer() ; Answer inbound calls exten = 201,2,Playback(silence/1) exten = 201,3,Background(simzy1) ; input an extension exten = 201,4,Wait(8) include = spa exten = 201,n,Hangup() [spa] exten =_201,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 t\ imes exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 t\ imes exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) exten = _2XX,3,HangUp() exten =_01,1,Dial(SIP/200) exten = 203,1,VoicemailMain exten = _2XX,1,Dial(SIP/${EXTEN},15) _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf HELP with dial plan!!
hello all, im looking for a way to do the following: when a SPECIFIC call comes through to asterisk through sip, i want it to b directed to a pool of specific sip extensions (9 extensions) where asterisk tries one after the other till lhe finds one of them thats actually on.i want to add a step for asterisk to follow which is, when a sip extension doesn't answer or its offline, instead of immediately transferring to voice mail, i want it to dial that sip holder's number so it transfers the call to his cellphone for example. and if he didn't answer his cellphone its then that i want it to direct it to voice mail.i want to add another item to the operator menu, instead of just receiving the call and telling the caller to either dial extension or 100 for operator, i want asterisk to offer the caller an additional option like for example pressing 2, would direct you to a list of key personnels with their respective extensions.please find below my extensions.conf: [sipura-line] exten = 201,1,Answer() ; Answer inbound calls exten = 201,2,Playback(silence/1) exten = 201,3,Background(simzy1) ; input an extension exten = 201,4,Wait(8) include = spa exten = 201,n,Hangup() [spa] exten =_201,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) exten = _2XX,3,HangUp() exten =_01,1,Dial(SIP/200) exten = 203,1,VoicemailMain exten = _2XX,1,Dial(SIP/${EXTEN},15) _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding funcionatlity to asterisk?! is it possible?!
hello all, im looking for a way to do the following: when a SPECIFIC call comes through to asterisk through sip, i want it to b directed to a pool of specific sip extensions (9 extensions) where asterisk tries one after the other till lhe finds one of them thats actually on.i want to add a step for asterisk to follow which is, when a sip extension doesn't answer or its offline, instead of immediately transferring to voice mail, i want it to dial that sip holder's number so it transfers the call to his cellphone for example. and if he didn't answer his cellphone its then that i want it to direct it to voice mail.i want to add another item to the operator menu, instead of just receiving the call and telling the caller to either dial extension or 100 for operator, i want asterisk to offer the caller an additional option like for example pressing 2, would direct you to a list of key personnels with their respective extensions.please find below my extensions.conf: [sipura-line] exten = 201,1,Answer() ; Answer inbound calls exten = 201,2,Playback(silence/1) exten = 201,3,Background(simzy1) ; input an extension exten = 201,4,Wait(8) include = spa exten = 201,n,Hangup() [spa] exten =_201,1,GoTo(sipura-line,${EXTEN},1) exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) exten = _1XX,3,HangUp() exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) exten = _2XX,3,HangUp() exten =_01,1,Dial(SIP/200) exten = 203,1,VoicemailMain exten = _2XX,1,Dial(SIP/${EXTEN},15) _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls not being answered by asterisk
Hey thanks for the help :) though i already did that, and the sip debugging info shows me tht its ringing on the respective sip extension (1002) but nothing is happening.. so i guess its true it IS a diala plan issue tht i am yet to figure it out ... Date: Sat, 24 May 2008 14:20:45 +0100 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk The first thing to do is type sip debug on the console and place the call from the Sipura. If you get a bunch of SIP messages flashing down your console you know the call is reaching Asterisk and it's most likely going to be an issue authenticating the call or a problem in your dial plan. If no SIP messages flash up then the call is not reaching your Asterisk server. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls not being answered by asterisk
Hello all, ive got the following setup currently: __Sipura 3102-PSTN | Lan | | |__asterisk i configured both asterisk and pstn to be able to receive/make calls through each other using sip of course.. but the thing is i want asterisk that when it receives an incoming call from sipura, to answer it, play msg that i recorded and wait for the caller to dial in an extension, where it would transfer the caller to that exntension, and in case the extension owner isnt available to answer it would direct him to his voicemail(tht i dont know how to set yet), and in case the caller didnt dial any extension in a certain amount of time, it automaticly directs it to a specific extensions i'd specify.. i tried the examples given in lots of forums and so on none of them worked, the phone keeps on ringing with every incomign dial plan ive specified without asterisk answering it.. the thing i did is that sipura directs incoming calls to 1002, so ive set the context of 1002 in sip.conf to a dial plan of [incoming-sipura] and ive set the commands i mentioned earlier tht i took out of those forums.. but theyre not working!!! anyone has an example i could go on with ? any help would be apreciated:) _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Roberto, i added this exntesions.conf [spa] Exten = _1XXX,1,Dial(SIP/${EXTEN}) exten = _0.,1,Dial(SIP/1009/${EXTEN:1}) and in sip.conf: [1009] username=1009 type=friend secret=1234 host=dynamic canreinvite=yes context=spa disallow=all allow=alaw dtmfmode=info qualify=yes callgroup=1 pickupgroup=1 which i have the extension 1009 in sip.conf directed to it.. and then tried calling out but it still gave me the same error! -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f0b840, SIP/1009/0144) in new stack -- Called 1009/0144 -- Got SIP response 503 Service Unavailable back from 192.168.0.111 -- SIP/1009-0821d888 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f0b840' status is 'CONGESTION' From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 21 May 2008 10:29:21 -0700 Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) Ciao Roland your dialplan:Exten = _1XX,1,Dial(SIP/${EXTEN}) _1XX is a three (3) digit number starting with 1, I'm not sure what happens if you dial 1009 but it seems that it is dialing. Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that extension which doesn't exist. your dial out should look something like: [outgoing] exten = _9.,1,Dial(SIP/100/${EXTEN:1}) where you're specifying that all the calls that starts with 9 will go to extension 100 (assuming that is your spa-3102) and there you dial the number dialed from the caller stripped by the 9 (that is the :1 after EXTEN)Now 9 is standard in USA for outside line, in some other countries is 0, you choose CiaoRoberto On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote: Hello Roberto, first of all, id like to thank you for your help with this.. secondly, i tried the configuration you gave me but it still gave me the same error..! but just to b sure ill tell u wht im doing.. after following ur advice to the letter.. i kept my asterisk configuration the same the only thing i edited in sip.conf is adding the port for the pstn extension to match the one in sipura 3102.. and gave the PSTN line interface on sipura the user id of 1009 then i called from my softphone 1009 so i could dial out.. and it gave me this error in asterisk cli: Connect attempt from '127.0.0.1' unable to authenticate -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f0e828, SIP/1009) in new stack -- Called 1009 -- Got SIP response 503 Service Unavailable back from 192.168.0.111 -- SIP/1009-0821d888 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f0e828' status is 'CONGESTION' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found is that the right way of doing this?! do i call 1009 (pstn line user id) or wht! ps: could us hare with me ur sip.conf and extensions.conf please just to compare mine with urs maybe something is missing! once again thanks for ur help :) Message: 22 Date: Wed, 21 May 2008 06:49:39 -0700 From: Roberto Milani [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252 Hi Roland I have 2 linksys spa-3102 working pretty good both dialing in and out and I followed this instructions to set it up: update to the latest firmware then: ..Go to the first tab ?Voice? and sixth sub-tab ?Line 1? SIP Settings: ..SIP Port: Notice that it is set to 5060 for line 1 and 5061 for PSTN Line (next tab). These port values must be correctly transferred to the correct contexts in sip.conf. Proxy and registration: ..Proxy: 192.168.5.70 The IP address of your Asterisk server Subscriber Information: ..Display Name: LivingRoom This will be the test phone, but any name would do as lone as it is used in the configuration files. ..User ID: LivingRoom ..Password: SomePassword ..Auth ID: LivingRoom probably not needed Dial Plan: ..Dial Plan: (*xx|[3469]11|0|00|[2-9]x| 1xxx[2-9]xS0|.) We have 10 digit local dialing. The default is set for seven digit local dialing. Adjust as needed. ..Emergency Number: Hmmm, I don?t know what to do here: it?s probably important, but it is poor form to dial 911 just to test. . . Help? Click Submit All Changes ..Go to the first tab ?Voice? and seventh sub-tab ?PSTN?: SIP Settings: ..SIP Port: Notice that it is set to 5061 for PSTN User and 5060 for Line 1. These port values must be correctly transferred to the correct contexts in sip.conf. Proxy and Registration: ..Proxy: 192.168.5.70
[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hello all, its been a while im trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones on pcs in my home.. i've set up 5 SIP extensions in sip.conf and made the dialing plan in extensions.conf.. i could make calls from 1 sip phone to another in my home.. but i cant call out using pstn line interface nor recieve calls.. please find below my topology as well as config info: (192.168.0.0) LAN__ || | softphone asterisk sipura-PSTN LINE Configuration: ASTERISK: sip.conf [101] type=peer port=5062 host=dynamic secret=1234 context=spa [103] type=peer port=5061 host=dynamic secret=1234 context=spa [100] type=peer port=5061 host=dynamic secret=1234 context=spa [111] type=peer port=5060 host=dynamic secret=1234 context=spa == === EXTENSIONS.CONF [spa] Exten = _1XX,1,Dial(SIP/${EXTEN}) == === and this is the settings i have right now for sipura 3102 in my PSTN LINE: http://img84.imageshack.us/my.php?image=40541922um2.jpg http://img98.imageshack.us/my.php?image=55448347ss9.jpg http://img262.imageshack.us/my.php?imag ... 472qz3.jpg ps: i read so many tutorials and none seems to help.. lately whenever i try to call out using my sipphone.. it gives me 503 service unavailable and this is wht shows on the CLI of asterisk when i set sip debug on.. ubuntu-pbx-desktop*CLI == Connect attempt from '127.0.0.1' unable to authenticate-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f05600, SIP/1009) in new stack-- Called 1009*CLI-- Got SIP response 410 Gone back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION' _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
this in some Google search. I don?t know what it does, but stuff seems to work. Help? FXO Timer Values (sec): ..PSTN Answer Delay: 5 Delay so that you can get the CID data. NghtShd at http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html claims that 5 seconds is long enough. Click Submit All Changes Ciao Roberto On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote: Hello all, its been a while im trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones on pcs in my home.. i've set up 5 SIP extensions in sip.conf and made the dialing plan in extensions.conf.. i could make calls from 1 sip phone to another in my home.. but i cant call out using pstn line interface nor recieve calls.. please find below my topology as well as config info: (192.168.0.0) LAN__ | | | softphone asterisk sipura-PSTN LINE Configuration: ASTERISK: sip.conf [101] type=peer port=5062 host=dynamic secret=1234 context=spa[103] type=peer port=5061 host=dynamic secret=1234 context=spa [100] type=peer port=5061 host=dynamic secret=1234 context=spa [111] type=peer port=5060 host=dynamic secret=1234 context=spa == === EXTENSIONS.CONF [spa] Exten = _1XX,1,Dial(SIP/${EXTEN}) == ===and this is the settings i have right now for sipura 3102 in my PSTN LINE: http://img84.imageshack.us/my.php?image=40541922um2.jpg http://img98.imageshack.us/my.php?image=55448347ss9.jpg http://img262.imageshack.us/my.php?imag ... 472qz3.jpg ps: i read so many tutorials and none seems to help.. lately whenever i try to call out using my sipphone.. it gives me 503 service unavailable and this is wht shows on the CLI of asterisk when i set sip debug on.. ubuntu-pbx-desktop*CLI == Connect attempt from '127.0.0.1' unable to authenticate -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f05600, SIP/1009) in new stack -- Called 1009*CLI -- Got SIP response 410 Gone back from 192.168.0.111 -- SIP/1009-081741d0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION' Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080521/7c9ef721/attachment.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 46, Issue 69 ** _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
yes thats the only thing i have in extensions.conf should there be anything else?! Message: 21Date: Wed, 21 May 2008 09:40:26 -0400From: Matt Watson [EMAIL PROTECTED]Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED]Content-Type: text/plain; charset=us-ascii Does your extensions.conf have any more configuration than what you've shown? If not, then you are lacking dialplan for anything but internal calls. --Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNdSent: Wednesday, May 21, 2008 9:01 AMTo: [EMAIL PROTECTED]: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) Hello all, its been a while im trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones on pcs in my home..i've set up 5 SIP extensions in sip.conf and made the dialing plan in extensions.conf..i could make calls from 1 sip phone to another in my home.. but i cant call out using pstn line interface nor recieve calls..please find below my topology as well as config info: (192.168.0.0) LAN__ | | |softphone asterisk sipura-PSTN LINE Configuration: ASTERISK: sip.conf [101]type=peerport=5062host=dynamicsecret=1234context=spa [103]type=peerport=5061host=dynamicsecret=1234context=spa [100]type=peerport=5061host=dynamicsecret=1234context=spa [111]type=peerport=5060host=dynamicsecret=1234context=spa == === EXTENSIONS.CONF [spa]Exten = _1XX,1,Dial(SIP/${EXTEN}) == === and this is the settings i have right now for sipura 3102 in my PSTN LINE: http://img84.imageshack.us/my.php?image=40541922um2.jpghttp://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg http://img98.imageshack.us/my.php?image=55448347ss9.jpghttp://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg http://img262.imageshack.us/my.php?imag ... 472qz3.jpghttp://img262.imageshack.us/my.php?imag%20...%20472qz3.jpg ps: i read so many tutorials and none seems to help..lately whenever i try to call out using my sipphone.. it gives me 503 service unavailable and this is wht shows on the CLI of asterisk when i set sip debug on.. ubuntu-pbx-desktop*CLI == Connect attempt from '127.0.0.1' unable to authenticate-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1003-b5f05600, SIP/1009) in new stack-- Called 1009*CLI-- Got SIP response 410 Gone back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1003-b5f05600' status is 'CONGESTION' _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Jose, i just did that, doesnt seem to work.. its still giving me the same error Date: Wed, 21 May 2008 11:02:36 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) I was seeing your print screen images, and the observation is. You are not doing any sip registration on the server since your Register option in the Tab PSTN Line is set to NO. you should change it to yes. (or add in the sip.conf the host=SPA_ip instead of dynamic). regards. -- Jose Flores Galicia [EMAIL PROTECTED] BriefCode Code Based Training _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users