[asterisk-users] SIP 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has it's own context, in which I set the following:
;=in extensions.conf==
[from-customerX]
exten = _X.,1,Set(CDR(accountcode)=) ;Here I change the
accountcode depending on each customer
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[from-customerY]
exten = _X.,1,Set(CDR(accountcode)=)
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[a2billing]
exten = _X.,1,DeadAGI(a2billing.php|1)
exten = _X.,2,Hangup(34)
;=
A2Billing authenticates and routes the call properly, but when the
termination gateway for the destination dialed by the customer rejects
the call, my Asterisk box sends 603 Declined to the customer.
It also happens when A2Billing doesn't find any route for that
destination, in which it should return 404 Not Found, but returns 603
Declined instead.
I tried to force every rejected attempt with 503 Service Unavailable
putting the Hangup(34) you see on my config, but it never seems to get
there.
The last thing I see on CLI running in verbose is: -- AGI Script
a2billing.php completed, returning 0
Is there anything I could do to return a different cause than 603
Declined?
I posted the same question on A2Billing's forum, but had no luck.
Thanks in advance,
Alejandro Mejia
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Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Anyone? Please advice. Thank you. On Sun, May 8, 2011 at 8:59 AM, GNUbie gnu...@gmail.com wrote: Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER to my Asterisk box. Below are the snippets of my Asterisk and SNOM 300 configurations including the logs for your reference. I hope anyone from this community can help me solve this problem. A HOWTO of a similar scenario will help a lot. Thank you in advance. Regards, GNUbie - - - ASTERISK v1.8.3.3 - - - [ /etc/asterisk/sip.conf ] [general] ... ... tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem tlscipher=ALL tlsclientmethod=tlsv1 tlsbindport=5061 externtlsport=5061 externtcpport=5061 tcpbindaddr=0.0.0.0 tcpbindport=5061 tcpenable=yes srvlookup=yes [361] username=361 secret=*** callerid=361-tls361 mailbox=361@family context=family transport=tls port=5061 type=friend host=dynamic dtmfmode=rfc2833 canreinvite=no nat=yes qualify=yes autoframing=yes encryption=yes *CLI core show version Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-04-22 17:50:44 UTC *CLI sip show settings Global Settings: UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: 0.0.0.0:5060 TLS SIP Bindaddress: 0.0.0.0:5061 Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No SIP domain support: Yes Realm. auth: No Our auth realm pbx.domain.com Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk rocks! SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 6 ms Q.850 Reason header: No Network QoS Settings: --- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: Yes Jitterbuffer forced: No Jitterbuffer max size: 200 Jitterbuffer resync: 1200 Jitterbuffer impl: fixed Jitterbuffer log: No Network Settings: --- SIP address remapping: Enabled using externhost Externhost: pbx.domain.com externaddr: 11.22.33.44:0 Externrefresh: 10 Localnet: 192.168.101.0/255.255.255.0 Global Signalling Settings: --- Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc) Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 15 RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 1800 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: not set Session Timers: Refuse Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 3000 Timer T1 minimum: 100 Timer B: 192000 No premature media: Yes Max forwards: 70 Default Settings: - Allowed transports: UDP Outbound transport: UDP Context: default Force rport: No DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk *CLI sip show peer 361 * Name : 361 Secret : Set MD5Secret : Not set Remote Secret: Not set Context : family Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 361@family VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : 361-tls 361 MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text
Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
On Thu, May 19, 2011 at 3:19 AM, GNUbie gnu...@gmail.com wrote: Anyone? Please advice. Thank you. That's WAYY too much info for me to go through right now, and I don't know anything about TLS registration but what I would ask for is if you have the following lines in your sip.conf domain=IP/FQDN of your asterisk server:TLS port so in your case add the lines domain=pbx.domain.com:5061 and then do a sip reload So far, all problems I've had, have been solved because of this. At the end of your sip.conf add those lines and it should fix your problem. HTH -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] v1.8.4: Extension Not found in Context?
On Wed, May 18, 2011 at 9:39 PM, A E [Gmail] all.efor...@gmail.com wrote: On Wed, May 18, 2011 at 9:29 PM, Paul Belanger pabelan...@digium.comwrote: On 11-05-18 08:01 PM, A E [Gmail] wrote: boxb*CLI dialplan show Test [ Context 'Test' created by 'pbx_config' ] '' = 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Hangup() [pbx_config] -= 1 extension (3 priorities) in 1 context. =- But when the call comes into boxb from box a, on box b CLI I see the msg: NOTICE[1613]: chan_sip.c:21581 handle_request_invite: Call from 'boxA' to extension '' rejected because extension not found in context 'Test'. WHY?? Thanks :( Does the peer using 'boxA' dialplan include context 'Test'? You mean in its definition/declaration in sip.conf? yes. sip.conf in Box B looks like this: [boxA] type=peer host=10.0.3.5 context=Test disallow=all allow=ulaw allow=g722 allow=g729 dtmfmode=rfc2833 canreinvite=no insecure=port,invite Ok, this problem is fixed. Once again, it was the damn domain= line in sip.conf Since I was using a non-standard port i.e. 5062, just using, autodomain=yes doesn't help. One needs to explicitly specify the local address and bindport to be included. But the message in the console is misleading. I think I need to open a bug/issue about this. If I have a udpbindaddr = 10.0.3.6:5062, then autodomain keyword, should actually be smart enough to read that and auto-include the port specified (if specified). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic dialing + SMS
Hi Guys Using call files might be easiest. But I d also try out AGI scripting too. I ll be sure to call back if I require any help. For the sms bit,...let's say I want to send bulk sms to multiple mobile devices. Thanks a lot Regards Sent from my BlackBerry® smartphone from Vodafone -Original Message- From: Steve Edwards asterisk@sedwards.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 18 May 2011 06:46:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Automatic dialing + SMS On Wed, 18 May 2011, gadgetron...@gmail.com wrote: Does it mean Asterisk has no in-built applications for auto dialing. Asterisk is a telephony Erector Set*. You get to build what you want. All the pieces are there. What scripting language can easily and best be used for the AGI. Easy may not be best. 'Easiest' is the language you know best. Best depends on your needs. A scripting language like PHP may be easiest for you if you know that language. A compiled language like C may be best if you want to run a bazillion calls per second. You can execute xxx AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. *) http://en.wikipedia.org/wiki/Erector_set -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic dialing + SMS
hello: i think you can use php and get message from GUI and send by php AGI. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com To: asterisk-users@lists.digium.com From: gadgetron...@gmail.com Date: Thu, 19 May 2011 09:57:23 + Subject: Re: [asterisk-users] Automatic dialing + SMS Hi Guys Using call files might be easiest. But I d also try out AGI scripting too. I ll be sure to call back if I require any help. For the sms bit,...let's say I want to send bulk sms to multiple mobile devices. Thanks a lot Regards Sent from my BlackBerry® smartphone from Vodafone -Original Message- From: Steve Edwards asterisk@sedwards.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 18 May 2011 06:46:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Automatic dialing + SMS On Wed, 18 May 2011, gadgetron...@gmail.com wrote: Does it mean Asterisk has no in-built applications for auto dialing. Asterisk is a telephony Erector Set*. You get to build what you want. All the pieces are there. What scripting language can easily and best be used for the AGI. Easy may not be best. 'Easiest' is the language you know best. Best depends on your needs. A scripting language like PHP may be easiest for you if you know that language. A compiled language like C may be best if you want to run a bazillion calls per second. You can execute xxx AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. *) http://en.wikipedia.org/wiki/Erector_set -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.comwrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
we are in a production environment and cannot reboot. besides, these zombie processes appear minutes after asterisk starts taking calls. On Thu, May 19, 2011 at 8:59 AM, Satish Patel satish...@hotmail.com wrote: Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
Actually not sure if it is asterisk generating these zombies... i'm starting to believe it's the enswitch_routed daemon, anybody familiar with enswitch? On Thu, May 19, 2011 at 9:02 AM, vip killa vipki...@gmail.com wrote: we are in a production environment and cannot reboot. besides, these zombie processes appear minutes after asterisk starts taking calls. On Thu, May 19, 2011 at 8:59 AM, Satish Patel satish...@hotmail.comwrote: Sometime reboot does help. -- Sent from my iPhone On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com wrote: I'm sure it's not nagios. I'm not running check_sip and i'm running nagios' NRPE on several other machines that do not have asterisk running. On Wed, May 18, 2011 at 4:43 PM, Alex Balashov abalas...@evaristesys.com abalas...@evaristesys.com wrote: Are you sure it's Asterisk creating the zombie processes, not the check_sip pinger in Nagios? Nagios is extremely bad with high throughput and concurrency, and check_sip is a wrapper around 'sipsak', which means it takes the full Timer T1 * 64 to time out if the Asterisk server is truly not available (about ~30-32 sec). On 05/18/2011 04:40 PM, vip killa wrote: I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale
operation, so I configured A2Billing for not to answer the call nor play any
greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer has
it's own context, in which I set the following:
;=in extensions.conf==
[from-customerX]
exten = _X.,1,Set(CDR(accountcode)=) ;Here I change the accountcode
depending on each customer
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[from-customerY]
exten = _X.,1,Set(CDR(accountcode)=)
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[a2billing]
exten = _X.,1,DeadAGI(a2billing.php|1)
exten = _X.,2,Hangup(34)
;=
A2Billing authenticates and routes the call properly, but when the
termination gateway for the destination dialed by the customer rejects the
call, my Asterisk box sends 603 Declined to the customer.
It also happens when A2Billing doesn't find any route for that destination,
in which it should return 404 Not Found, but returns 603 Declined instead.
I tried to force every rejected attempt with 503 Service Unavailable
putting the Hangup(34) you see on my config, but it never seems to get there.
The last thing I see on CLI running in verbose is: -- AGI Script
a2billing.php completed, returning 0
Is there anything I could do to return a different cause than 603 Declined?
I posted the same question on A2Billing's forum, but had no luck.
Thanks in advance,
Alejandro Mejia
--
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[asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
Hi, I am trying to use ConfBridge application, but it throws Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) error. Please see console output below. -- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005, 1001) in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference bridge '1001' [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368 destroy_conference_bridge: Destroying conference bridge '1001' [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435 join_conference_bridge: Conference bridge '1001' could not be created. Could someone please let me know what is required to make it work? Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has it's own context, in which I set the following:
;=in extensions.conf==
[from-customerX]
exten = _X.,1,Set(CDR(accountcode)=) ;Here I change the
accountcode depending on each customer
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[from-customerY]
exten = _X.,1,Set(CDR(accountcode)=)
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)
[a2billing]
exten = _X.,1,DeadAGI(a2billing.php|1)
exten = _X.,2,Hangup(34)
;=
A2Billing authenticates and routes the call properly, but when the
termination gateway for the destination dialed by the customer rejects
the call, my Asterisk box sends 603 Declined to the customer.
It also happens when A2Billing doesn't find any route for that
destination, in which it should return 404 Not Found, but returns 603
Declined instead.
I tried to force every rejected attempt with 503 Service Unavailable
putting the Hangup(34) you see on my config, but it never seems to get
there.
The last thing I see on CLI running in verbose is: -- AGI Script
a2billing.php completed, returning 0
Is there anything I could do to return a different cause than 603
Declined?
I posted the same question on A2Billing's forum, but had no luck.
Thanks in advance,
Alejandro Mejia
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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
On 11-05-19 09:39 AM, Chris Maciejewski wrote: Hi, I am trying to use ConfBridge application, but it throws Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) error. Please see console output below. -- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005, 1001) in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference bridge '1001' [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368 destroy_conference_bridge: Destroying conference bridge '1001' [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435 join_conference_bridge: Conference bridge '1001' could not be created. What version of Asterisk are you using? ConfBridge was rewritten in trunk and would be good to see if you have the same issue. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set moh without setting queue music with SetMusicOnHold
Hi Alex, dunno, it changes all moh (moh and queues music) on the channel, haven't tried with other people already in the queue I was told to stop testing when I found out I cannot achieve my goal. :| Giorgio Incantalupo On 05/18/2011 05:41 PM, Alex Balashov wrote: On 05/18/2011 11:34 AM, gincantalupo wrote: I could create 2 queues, one for italians and one for strangers calling but there is no point where you can change the moh except before executing the queue command but the queue moh changes as side-effect: Hmm. When you use SetMusicOnHold, does it change the queue MOH only for the particular member/channel that joins it, or for the entire queue globally, for everyone already in the queue, etc? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager logged on/off messages
Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering
Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX in 1.8.4, going back to 1.8.3.3 everything works. I did open https://issues.asterisk.org/view.php?id=19264 and included a SIP trace. Sorry all, I did not follow up adequately. Definitely a problem with 1.6.2.18 and the issue # is 18951. Fixed in 1.8.3.3; Cisco 79xx registered fine. I don't know about 1.8.4 yet; haven't installed it for testing yet. Cassius This fix definitely not in 1.8.4; I also dropped back to 1.8.3.3 on a test box and Cisco 79XX's register correctly. Thanks for opening the issue; will check 1.8.5rc when it's available. Cassius On Fri, May 6, 2011 at 12:24 PM, Julian Lyndon-Smith aster...@dotr.com wrote: It was my problem ;) https://issues.asterisk.org/view.php?id=18951 fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com http://issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
What version of Asterisk are you using? ConfBridge was rewritten in trunk and would be good to see if you have the same issue. Hi Paul, I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661) and it still doesn't work, this time throwing error as below: -- Executing [501@services:3] ConfBridge(SIP/OpenSER-0001, 10001) in new stack [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:775 join_conference_bridge: Trying to find conference bridge '10001' [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:736 destroy_conference_bridge: Destroying conference bridge '10001' [May 19 16:11:58] ERROR[30778]: app_confbridge.c:814 join_conference_bridge: Conference bridge '10001' could not be created. Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager logged on/off messages
On 05/19/2011 12:05 PM, Ishfaq Malik wrote: Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Hi Ishfaq, I think that you might use a proxy, which connection is always active (see Astman Proxy), and send commands to it. It will not have to login/logoff everytime. Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping incompatible voice frame
Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager logged on/off messages
On Thu, 2011-05-19 at 12:15 -0400, Jose P. Espinal wrote: On 05/19/2011 12:05 PM, Ishfaq Malik wrote: Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Hi Ishfaq, I think that you might use a proxy, which connection is always active (see Astman Proxy), and send commands to it. It will not have to login/logoff everytime. Regards, Hi Thanks for that suggestion, I think it will help for additional reasons as well so cheers. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Static Vs Dynamic queue confusion
I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member = technology/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] click to call with php
Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping incompatible voice frame
For 2 different hosts. SIP/voxbone.com and SIP/4420 From: RSCL Mumbai Sent: Thu 5/19/2011 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call with php
You only need to tell your PHP script to write a .call file on /var/spool/asterisk/outgoing/ directory using the syntax described here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out I'm not a PHP programmer, so the PHP part is up to you hehe. There are other methods like using manager, but to keep it simple, I recommend you to use .call files. Good luck... On 19/05/2011 10:44 a.m., salaheddine elharit wrote: Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call with php
ok thank you i will test this solution and i will update you :) 2011/5/19 Alejandro Mejia Evertsz ame...@gua.net You only need to tell your PHP script to write a .call file on /var/spool/asterisk/outgoing/ directory using the syntax described here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out I'm not a PHP programmer, so the PHP part is up to you hehe. There are other methods like using manager, but to keep it simple, I recommend you to use .call files. Good luck... On 19/05/2011 10:44 a.m., salaheddine elharit wrote: Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping incompatible voice frame
But why does *our *native format keep changing :) Going by layman terms, if native format is alaw and someone speaks to me in uLaw, I will say *format changed*. But if native format is alaw and someone is talking with me in alaw, I should be happy. On Thu, May 19, 2011 at 10:28 PM, Terry Brummell te...@brummell.net wrote: For 2 different hosts. SIP/voxbone.com and SIP/4420 -- *From:* RSCL Mumbai *Sent:* Thu 5/19/2011 12:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-cpu utilization 60 %
Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 Anyone else facing high CPU usage problem with Asterisk 1.6.2.13 or any Elastix 2.0.3 users here ? With just 3 concurrent calls and none in queue, the CPU is constantly above 40%. The moment CPU goes above 50%, calls start to break. I am a newbie and at lack of options... Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping incompatible voice frame
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, May 19, 2011 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame But why does our native format keep changing :) Going by layman terms, if native format is alaw and someone speaks to me in uLaw, I will say format changed. But if native format is alaw and someone is talking with me in alaw, I should be happy. As far as I can tell this is a bug. I've also experienced similar issues with our 1.8 box, but this is a production box and not easy to gather the needed troubleshooting info. My solution is to make sure no transcoding is going on. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP335 3.3.1 Call Waiting
I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with 3.2.4 it would automatically show call waiting name and number without pressing any keys. It could be possible I missed a setting, but I didn't see anything in the admin guide. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pridialplan/ prilocaldialplan
Hi I change the chan_dahdi.conf and restart dahdi: prilocaldialplan=private pridialplan=private But, in debug i see the following informations: 1 Calling Number (len= 8) [ Ext: 0 TON: *National Number (2) * NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1570' ] 1 [70 0a 80 30 38 31 37 34 37 39 35 36] 1 Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '81747956' ] I set Private TON, but display National TON. Thank's Att, Rafael Saraiva 2011/5/19 Захаров Антон ins...@mail.ru Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-cpu utilization 60 %
How much memory have allocate to VM ? and send top or ps command output. Date: Thu, 19 May 2011 22:44:58 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 Anyone else facing high CPU usage problem with Asterisk 1.6.2.13 or any Elastix 2.0.3 users here ? With just 3 concurrent calls and none in queue, the CPU is constantly above 40%. The moment CPU goes above 50%, calls start to break. I am a newbie and at lack of options... Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static Vs Dynamic queue confusion
agents.conf agent = 7101,1234,Agent1 agent = 7102,1234,Agent2 queues.conf ... ... member = Agent/7201 member = Agent/7202 CLI output holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet Agent/7101 with penalty 1 (dynamic) (Unavailable) has taken no calls yet Agent/7102 with penalty 1 (dynamic) (Unavailable) has taken no calls yet No Callers agents are not getting calls. and what is Invalid ? From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 19 May 2011 16:41:02 + Subject: [asterisk-users] Static Vs Dynamic queue confusion I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member = technology/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
On 11-05-19 12:13 PM, Chris Maciejewski wrote: What version of Asterisk are you using? ConfBridge was rewritten in trunk and would be good to see if you have the same issue. Hi Paul, I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661) and it still doesn't work, this time throwing error as below: -- Executing [501@services:3] ConfBridge(SIP/OpenSER-0001, 10001) in new stack [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:775 join_conference_bridge: Trying to find conference bridge '10001' [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:736 destroy_conference_bridge: Destroying conference bridge '10001' [May 19 16:11:58] ERROR[30778]: app_confbridge.c:814 join_conference_bridge: Conference bridge '10001' could not be created. Attach a debug[1] log so we can see what is happening. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
perhaps you forgot to run make config _after_ installing dahdi drivers --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: isr...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 May, 2011 3:48:05 PM Subject: Re: [asterisk-users] dahdi command not available Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
also, make sure that when you installed asterisk, the option to load the dahdi module was select. when you run a ./configure it scans your system and when you run make menuselect, the resource module dahdi will be marked to be compiled and installed :) --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: isr...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 May, 2011 3:48:05 PM Subject: Re: [asterisk-users] dahdi command not available Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
Thanks for reply Marcelo, I don't know what was the problem but after reboot machine it works! I am pretty sure i did service dahdi start/stop but that didn't work. -S Date: Thu, 19 May 2011 16:44:18 -0300 From: ellm...@freeddom.com To: isr...@gmail.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi command not available also, make sure that when you installed asterisk, the option to load the dahdi module was select. when you run a ./configure it scans your system and when you run make menuselect, the resource module dahdi will be marked to be compiled and installed :) --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: isr...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 May, 2011 3:48:05 PM Subject: Re: [asterisk-users] dahdi command not available Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
I'm glad you got it right! :) cheers, --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: satish patel satish...@hotmail.com To: asterisk-users asterisk-users@lists.digium.com Sent: Thursday, 19 May, 2011 5:13:03 PM Subject: Re: [asterisk-users] dahdi command not available Thanks for reply Marcelo, I don't know what was the problem but after reboot machine it works! I am pretty sure i did service dahdi start/stop but that didn't work. -S Date: Thu, 19 May 2011 16:44:18 -0300 From: ellm...@freeddom.com To: isr...@gmail.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi command not available also, make sure that when you installed asterisk, the option to load the dahdi module was select. when you run a ./configure it scans your system and when you run make menuselect, the resource module dahdi will be marked to be compiled and installed :) --- Marcelo Ellmann Freeddom Tecnologia e Serviços S/A +55 11 52133200 Ramal 1016 - Original Message - From: isr...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 May, 2011 3:48:05 PM Subject: Re: [asterisk-users] dahdi command not available Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent (Invalid) has taken no calls yet
How to get rid on following.. why its Invalid ? holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: FW: realtime mysql - p4]
Ok, i tried the suggestion: Instead of: sippuser = resource, database_name, table_name sippeer = resource, database_name, table_name I put in: sippuser = resource, context, table_name sippeer = resource, context, table_name Unfortunately, with the same results. btw i tried both general as default Besids the commands i tried below, isn't there any other way to see what's going on? Perhaps it is totally unrelated, but if i perform a mysql-login on the prompt, i first have to select the database manualy, ie it isn't selected by default for the created mysqluser [in this case: voipadmin] Other wild idea, is there a minimum number of fields that haved to be filled? And why is asterisk complaining about not being able to find the databse, when trying to fill it from the asterisk-CLI? My database _is_ named asterisk.. kc3054*CLI realtime update sipusers set SET port = 4343 WHERE name = 0277611 Failed to update. Check the debug log for possible SQL related entries. Command 'realtime update sipusers set SET port = 4343 WHERE name = 0277611' failed. [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql: MySQL RealTime: Invalid database specified: 'asterisk' (check res_mysql.conf) I mean, is that silly or what? # grep mysql extconfig.conf |grep sip ;sipusers = mysql,asterisk,sip_devices ;sippeers = mysql,asterisk,sip_devices ;sipusers = mysql,general,sip_devices ;sippeers = mysql,general,sip_devices sipusers = mysql,default,sip_devices sippeers = mysql,default,sip_devices kc3054*CLI module show like mysql Module Description Use Count cdr_mysql.so MySQL CDR Backend0 res_config_mysql.soMySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 3 modules loaded kc3054*CLI kc3054*CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheimdefault No Yes 027761125b06d3a0b5ef73 default No Yes kc3054*CLI kc3054*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Realtime 0277611(Unspecified)D N 0Unmonitored j.witvliet (Unspecified)D N 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] kc3054*CLI kc3054*CLI kc3054*CLI kc3054*CLI realtime mysql cache kc3054*CLI realtime mysql status general connected to asterisk@127.0.0.1, port 3306 with username voipadmin for 18 seconds. kc3054*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call with php
I had issue with call files. They would lock up the system (this was 5 years ago so maybe things have changed.) - Original Message - From: Alejandro Mejia Evertsz To: asterisk-users@lists.digium.com Sent: Thursday, May 19, 2011 19:58 Subject: Re: [asterisk-users] click to call with php You only need to tell your PHP script to write a .call file on /var/spool/asterisk/outgoing/ directory using the syntax described here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out I'm not a PHP programmer, so the PHP part is up to you hehe. There are other methods like using manager, but to keep it simple, I recommend you to use .call files. Good luck... On 19/05/2011 10:44 a.m., salaheddine elharit wrote: Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent (Invalid) has taken no calls yet
If you go for 1.8,Don't read from http://www.asteriskguru.com/tutorials/queues.html. It is bit backdated information. Rather I would suggest you to check http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html. Queue members are considered INVALID, if their device status is Invalid. This is somewhat an error condition.SIP channels are the only type that provide true device state information. I also suggest you to read 'The agents.conf File' section from given link for more information. [SATISH] On Fri, May 20, 2011 at 2:40 AM, satish patel satish...@hotmail.com wrote: How to get rid on following.. why its Invalid ? holler*CLI queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call with php
If you don't like callfiles, another option is AMI. Check the sample code from http://tycoontalk.freelancer.com/php-forum/156207-click-to-call-using-php.html, do some changes as per your requirements. I would love to use callfiles as it gives more flexibility(as per my understanding) compared to AMI. [SATISH] On Fri, May 20, 2011 at 9:55 AM, Dovid Bender asteriskus...@dovid.netwrote: I had issue with call files. They would lock up the system (this was 5 years ago so maybe things have changed.) - Original Message - *From:* Alejandro Mejia Evertsz ame...@gua.net *To:* asterisk-users@lists.digium.com *Sent:* Thursday, May 19, 2011 19:58 *Subject:* Re: [asterisk-users] click to call with php You only need to tell your PHP script to write a .call file on /var/spool/asterisk/outgoing/ directory using the syntax described here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out I'm not a PHP programmer, so the PHP part is up to you hehe. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using a feature from AMI or CLI
Hi, I've defined a feature using a macro in features.conf : special = #2,peer,Macro,special Everything is working if the user use the phone key. But i would like to call the feature (or the Macro on the peer channel) from AMI or CLI. First i thought i would be simple, but i did not find any solution. Does someone has an idea ? Thank you very much. Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's zombie processes
On 19/05/11 16:04, vip killa wrote: Actually not sure if it is asterisk generating these zombies... i'm starting to believe it's the enswitch_routed daemon, anybody familiar with enswitch? Hello, I am the lead developer of Enswitch. Enswitch comes with commercial support as standard, so if you suspect there's a problem with Enswitch we (or our partners if you've bought a system through them) would be delighted to take a look as part of normal support. If you're unsure of how to do this, please drop me an email off-list letting me know your name and what company you work for and I can give you details of how to contact support. Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pridialplan/ prilocaldialplan
Yeap, I couldn't set Private TON too. Try to set all _prefix variables in chan_dahdi.conf and use dynamic prilocaldialplan. On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote: Hi I change the chan_dahdi.conf and restart dahdi: prilocaldialplan=private pridialplan=private But, in debug i see the following informations: 1 Calling Number (len= 8) [ Ext: 0 TON: *National Number (2) * NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1570' ] 1 [70 0a 80 30 38 31 37 34 37 39 35 36] 1 Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '81747956' ] I set Private TON, but display National TON. Thank's Att, Rafael Saraiva 2011/5/19 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
