Re: [asterisk-users] Asterisk Installation
Hey Chris, Starts from here, https://wiki.asterisk.org/wiki/display/AST/Getting+Started or try Asterisk Complete guide in pdf format. If you are looking for something graphical, go for elastix or freepbx. Thanks ~Arun On Thu, Nov 24, 2016 at 12:28 AM, christopher kamutumwa < chriskamutu...@gmail.com> wrote: > Goodday users > > Am quite new to asterisk and trying to configure it with an fxo and fxs > digium card. also i need a gui interface implemented. I have a centos 6.8 > server any tutorial i could use for install and configuration? would > appreciate. > > Thanks > > Chris > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice clarity issue
Hello all, Im using a GSM gateway device for making outbound calls. GSM device is connected to one of my SIP peer. Now am facing a lot of voice signal problems. I checked with my vendor and there is no issues with signal and device. Any settings in asterisk? Thanks Arun -- _ -- 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] Attack on Sip server.
Hi, Change the protocol from tcp to udp in iptables. ~Arun On 27 Jun 2014 20:07, Anurag Rana anuragrana31...@gmail.com wrote: Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the packets from attacker contains string 'VaxSIPUserAgent/3.1' ) but it failed. Packets are still flowing in. iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent --algo bm -j DROP Its something like this Registration from '30 sp:30@my_public_ip:5060 failed for '192.168.xxx.xxx:6373' - Wrong Password and there are approx 10 request per minute of this type. Please suggest some way to stop this. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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] T1 Card RED ALARM
Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun -- _ -- 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] T1 Card RED ALARM
Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com wrote: On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com wrote: Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun It could still be some sort of system config issue, even if you think everything is configured the same. Have you tried moving the T1 card from the Bad system to the good system? That will at least help narrow down if it's a bad card / port, or a config issue. -Josh -- _ -- 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] T1 Card RED ALARM
Cables are workig fine in my other box. On 25 Jun 2014 00:46, Steve Totaro stot...@totarotechnologies.com wrote: Remember to always check your cables first. Thanks, Steve T On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote: Thank you Josh for your valuable reply. I will do try changing the server and let you know what happening. ~Arun On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com wrote: On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com wrote: Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured Thanks ~Arun It could still be some sort of system config issue, even if you think everything is configured the same. Have you tried moving the T1 card from the Bad system to the good system? That will at least help narrow down if it's a bad card / port, or a config issue. -Josh -- _ -- 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] T1 Card RED ALARM
Its an old box with Asterisk 1.2 On 25 Jun 2014 03:46, Mc GRATH Ricardo mcgra...@mail2web.com wrote: Why you configure zaptel.conf? should configure on dahdi files Mc GRATH Ricardo E-Mail mcgra...@mail2web.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
[asterisk-users] My new blog http://cciev.ciscovoicetech.com/
Hi Guys, Soon, I'll be starting a new section related to Asterisk (around 4 years of full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*) so let me know if you like to see some topic coming. Cheers Arun -- _ -- 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 CCM, CME Integration
HI All, I got solved this issue. Thanks all for your help Arun On Sun, May 24, 2009 at 1:58 AM, David Backeberg dbackeb...@gmail.comwrote: On Wed, May 20, 2009 at 12:44 AM, Arun Kumar arunv...@gmail.com wrote: here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see 500 Internal Server Error back from my CCM IP and getting standard asterisk message saying all circuits are busy now . as per my understanding it should go to my cue. You need to enable better debugging on the Cisco side. You shouldn't be getting a 500 internal server error. You need to debug the Cisco and find out what it says besides 500 internal server error. There should be logging for an error like that. ___ -- 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] Fwd: Asterisk CCM, CME Integration
Hi All, please provide some help. I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones Asterisk SIP Trunk Call Manager - CME - 461X Phones 461X Phones CME - my dial peer points to Asterisk IP for 600X Phones so in the above setup I'm able to call from Asterisk to my CME and vice-versa. here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see 500 Internal Server Error back from my CCM IP and getting standard asterisk message saying all circuits are busy now . as per my understanding it should go to my cue. please advise and let me know if you need any other details. Regards Arun ___ -- 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 CCM, CME Integration
Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones Asterisk SIP Trunk Call Manager - CME - 461X Phones 461X Phones CME - my dial peer points to Asterisk IP for 600X Phones so in the above setup I'm able to call from Asterisk to my CME and vice-versa. here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see 500 Internal Server Error back from my CCM IP and getting standard asterisk message saying all circuits are busy now . as per my understanding it should go to my cue. please advise and let me know if you need any other details. Regards Arun ___ -- 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] about the Dial application
Hi guys I am working in Kanpur, India. When someone calls to my server i forward the call to someone else by Dial command. After dialing it says Native bridging. And after that I am unable to detect whether the call was answered, the called number was busy or the call was not completed. One more issue, I want to record the discussion going on between the two persons. I used the options wW, but was unable to do it. Please if someone know how to solve these problems, please help me out. Thanks in advance.. arun From Chandigarh to Chennai - find friends all over India.. Go to http://in.promos.yahoo.com/groups/citygroups/___ -- 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] Fwd: Detection of Answer, hangup, busy etc while using Dial command
-- Forwarded message -- From: Arun Kumar Chaudhary [EMAIL PROTECTED] Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: [EMAIL PROTECTED] Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi script. I am unable to detect whether it is connected to the dialed number, if the call is picked up, if the called person disconnects the call, or the line is busy. On the asterisk CLI it show native bridging started. I also need to record the discussion. That is someone call my server. I connect him to someone else... and now I wand to record this discussion. Dial command uses option wW to record but it doesn't work. If anybody have answer of my question plz mail me. I will be really thnkful to you guys. Sincerely Arun Chaudhary ___ -- 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] Asterisk Nokia
Hi, I've two wifi-phones 1. Nokia e65 2. HP Ipaq I've configure two sip exten in my asterisk and using these exten in my phones. But my Nokia phone is keep on loosing the connectivity very soon life 1-2 min the qualify packet will be double of my HP. So, when I try to call my Nokia SIP exten it takes very long, but HP works fine. I tested one more phone also that works fine. so, I've a feeling that some kind of tweak is need with Nokia. thanks arun ___ --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] Softphone to be installed on the Mobile
try to use http://www.fring.com/download/ On Nov 21, 2007 3:28 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --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] Need help in selecting DTMF Mode
Hi here is my setup : 1. USER - PSTN - Asterisk A - IAX2 Trunk - Asterisk B - SER - Asterisk C (Accepting DTMF) All Asterisk box has dtmfmode = inband, when user pressed DTMF able to receive and working fine. 2. Asterisk C --- Dial Customer Customer input DTMF and its not taking any dtmf but If I change dtmfmode to auto Asterisk C will take DTMF from users but my first Scenario fails if I change dtmfmode = auto in Asterisk C. Need urgent help. Thanks Arun ___ --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] DTMF Problem
Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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] SER with Asterisk intergration
just configure SER on another port and use. On 11/1/07, satish patel [EMAIL PROTECTED] wrote: Dear all anybody have implement SER with Asterisk in single machine ?? i have asterisk with 200 SIP device but i voice qulity and load of asterisk is bit high so i need to implement SER for SIP registra and asterisk for feature Rgerads PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] flooded by Maximum trunk data space exceeded messages
try to reduce number of calls on trunk or create multiple trunks. On 10/31/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, Using 1.4.13 and trunking a single iax channel to a similar box my asterisk console is flooded with: [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space exceeded to xx.xx.xx.xx:4569 Known issue? ___ --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] Asterisk H323 Config
Hi Need help on this setup: Incoming DID in H323 Asterisk Server -- SIP Phone please tell me to achieve this above setup what needs to be done in Asterisk. thanks Arun ___ --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 Voicemail
Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] RTP Call Disconnect
Hi All, UA Asterisk Server - UB if there is no rtp for a specified number of minutes / seconds then I want to disconnect the call. I've tried using rtptimout and rtpholdtimeout but no luck pls guide. thanks arun ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] How to verify IAX trunking
run iax2 show peers and see next to port (T) if it comes then you are using IAX2 Trunking feature. On 8/10/07, George Pajari [EMAIL PROTECTED] wrote: How can one verify that IAX trunking is in effect and that Asterisk is trunking multiple call paths between two Asterisk servers? With 1.4.10 on both ends, entering iax2 set debug trunk on either end merely results in the response: IAX2 Trunk Debug Requested and nothing more. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --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] Asterisk-1.2.22 DeadAGI Hangup
Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun ___ --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 Freeze
HI Here is my info: Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my agents are not able to hear any audio only solution is to restart the whole box. Please advise soon. thanks arun ___ --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 PRI Busy Problem
issue is got solved by moving to another pri card and now congestion works fine with my ISP. thanks all. On 7/18/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: On 7/17/07, Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote: I did a quick test. What happens is Congestion() answers the channel and leaves it open. IE do a 'show channels' and you will see the channel is still open on your end. What happens in you pass a timeout to the Congestion() application, and then hangup the call after that, as show below? exten = 4340,15,Queue(test,rt,,,10) exten = 4340,16,Congestion(3) exten = 4340,17,Hangup() Give that a try and see if it helps. -- Jared Smith Community Relations Manager Digium, Inc. Yes, that seems to solve Arun's problem. When I do Congestion(1) I receive approx 1 second of congestion tone and then ringing and the SprintPCS message We are unable to complete your call at this time (because I call into it from a Sprint mobile phone) ___ --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] Asterisk PRI Busy Problem
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my provider says he is not getting any busy signal from my side and
he says for all incoming numbers (30) he is getting back only one number
from asterisk box(4340).
here is my dial plan for one incoming DID:
exten = 4340,1,GotoIfTime(*|*|25|dec?ccagents,4340,6)
exten = 4340,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,4340,7)
exten = 4340,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,4340,7)
exten = 4340,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,4340,7)
exten = 4340,5,GotoIfTime(09:00-20:00|mon-sun,*,*?ccagents,4340,7)
exten = 4340,6,Goto(out-of-hours,5001,1)
exten = 4340,7,Set(GROUP(${EXTEN})=MAX_CALLS)
exten = 4340,8,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])} 3]?16)
exten = 4340,9,Set(GROUP(${CALLERIDNUM})=MAX_CALLS)
exten = 4340,10,Answer()
exten = 4340,11,Playback(custom/next-avail-advisor)
exten =
4340,12,Set(MONITOR_FILENAME=/var/spool/asterisk/q/tcarehwsupport-${TIMESTAMP}-${UNIQUEID})
exten = 4340,13,Monitor(wav,${MONITOR_FILENAME},mb)
exten = 4340,14,NoOp(${QUEUESTATUS})
exten = 4340,15,Queue(test,rt,,,10)
exten = 4340,16,Congestion()
zapata.conf:
---
[trunkgroups]
[channels]
language=en
context=ccagents
switchtype=euroisdn
pridialplan=unknown
overlapdial=yes
signalling=pri_cpe
rxgain = 0.0
txgain = 0.0
usecallerid=yes
hidecallerid=yes
callerid=asreceived
callwaiting=yes
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
immediate=no
cidsignalling=v23
callwaitingcallerid=yes
priindication = outofband
resetinterval =
group = 1
channel = 1-15
channel = 17-31
group = 2
channel = 32-46
channel = 48-62
thanks
arun
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[asterisk-users] Early Media Handling
Hi using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call it should goto my specified extension. my php script: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); Please help thanks arun ___ --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 Help
Hi I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message . Please help . I've tried here is my code to place calls but in this I see no of failure calls are more than 50%. so please advise. $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); thanks arun ___ --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 Manager
Hi this is my code for * manager: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); when call gets answered it goes to my specified exten can I also handle if my call is not got answered b'coz of some reason. that is when get ans goto exten= 101 if call is not got and goto exten=102 please help. thanks arun ___ --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] Help with IAX Trunk
thanks for reply. I've same setup with siml. incoming calls 10-12 it works fine but some time my machies get hang and gives same IAX max data space error. thanks On 6/27/07, Jared Smith [EMAIL PROTECTED] wrote: On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote: so , how much bandwidth I need for 30 simul. calls ? If you're using IAX2 trunking, the bandwidth requirements will be much less than if you're not using IAX2 trunking. Make sure you have trunk=yes in the peer definition in iax.conf. Off the top of my head (without actually running the numbers), I would guess that 30 simultaneous calls using the g.729 codec and using IAX2 trunking would take less than 512kbit/sec in each direction. to support 30 calls over IAX2 do I've to change some setting during compile time or not ? No, just make sure you have a suitable timing source (Digium card, ztdummy, etc.) for the IAX2 trunk. -Jared ___ --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] Help with IAX Trunk
Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) UK === User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls over IAX2 do I've to change some setting during compile time or not ? pls suggest. thanks arun ___ --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] iax trunking on OpenBSD
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to use IAX2 trunk. On 6/7/07, Sebastian Reitenbach [EMAIL PROTECTED] wrote: Hi, do I have a chance to use iax trunking on OpenBSD where there is no zaptel driver or ztdummy available? Do I can use sth. else as timing source? kind regards Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Trunk No Sound
Hi I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk and they were working fine b'coz of bandwidth issue I changed from SIP to IAX now I'm facing a strange problem after some time on the cli of my asterisk box I see lots of messages of IAX2 trunk and b'coz of that my agents are not able to hear any thing and I've restart my * box. Please guide me what I do to resolve this issue. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Trunk No Sound
On 6/5/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Arun -
I've two boxes connected over IAX2 trunk before IAX I was using SIP
trunk
and they were working fine b'coz of bandwidth issue I changed from SIP
to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that my
agents are not able to hear any thing and I've restart my * box. Please
guide me what I do to resolve this issue.
1. How are these boxes connected (over the internet, on the same LAN,
etc)?
Connected over Internet between two offices
2. Can you post your iax.conf and relevant portions of extensions.conf?
Server A
-
[general]
bindport=4569
bandwidth=high
tos=lowdelay
register = user:[EMAIL PROTECTED]
[uk_trunk]
username=user
secret=pass
host=server B IP
qualify=no
context=default
type=friend
trunk=yes
disallow=all
allow=g729
[tcdubai]
type=friend
host=dynamic
trunk=yes
username=user
secret=pass
context=default
qualify=no
disallow=all
allow=g729
Server B
-
[general]
bindport=4569
bandwidth=high
tos=lowdelay
register = user:[EMAIL PROTECTED] A
[dubai_trunk]
username=user
secret=pass
host =server A IP
context=default
type=friend
trunk=yes
disallow=all
allow=g729
qualify=no
[tcuk]
type=friend
host=dynamic
trunk=yes
username=user
secret=pass
context=default
disallow=all
allow=g729
qualify=no
exten.conf on Server B (Incoming)
exten = _4XXX,1,Dial(IAX2/dubai_trunk/${EXTEN},,to)
exten.conf on Server A (Outbound)
---
exten = _0.,1,DeadAGI(
queueDial.agi|${EXTEN}|IAX2/uk_trunk/${EXTEN}|outbound|${CALLERID})
3. Can you post some of the CLI errors you mentioned?
iax2_trunk_queue: Maximum data space exceeded
and once this start it never gets stopped so I've to kill the asterisk and
restart the whole box. Instead of restart whole box if I just try to restart
the asterisk my agents not able to hear any voice.
thanks
arun
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[asterisk-users] Digium Card
HI I'm looking for a card that support both PRI and TDM. Please suggest me ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Trunk Problem
Hi I've two boxes connected over IAX2 trunk but suddenly my cli is getting flood with these messages: iax2_trunk_queue: Maximum data space exceeded and b'coz of that my agents are not able to hear any thing. when this happened that time there were 9 calls. my * version is 1.2.18 and 1.2.14 thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 License
HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Crash
Hi I've two boxes connected via IAX2 Trunk were working fine from few days suddenly today one box is got crashed with this message 2007-06-03 12:25:37 WARNING[26511]: chan_sip.c:2612 sip_write: Can't send 4113608 type frames with SIP write my version of * is 1.2.14 on FC4 thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue
HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue
Hi my * box is giving me these warning and b'coz of second warning line my agents are not able to hear the announcement in the queue some time it happen many time 2007-06-03 13:40:30 WARNING[28016]: chan_sip.c:2612 sip_write: Can't send 4113568 type frames with SIP write 2007-06-03 13:40:30 WARNING[28016]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue
Hi sorry for not details. when ever I see this message on * console my agents are not able to listen to announcement. thanks arun On 6/3/07, Mattt [EMAIL PROTECTED] wrote: And you don't find that sufficiently self-explanatory? On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote: HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs Cheers, Mattt. - ROMATel - VoIP made easy - http://romatel.net - SpotSafe - WiFi Hotspot solution - http://spotsafe.net There are only 10 kinds of people. Those who understand binary, and those that don't... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: TC400B load problem
-- Forwarded message -- From: Arun Kumar [EMAIL PROTECTED] Date: May 13, 2007 5:40 PM Subject: TC400B load problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 please help thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: TC400B load problem
thanks Matthew, I'll try to call Digium. On 5/14/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: On May 14, 2007, at 4:53 AM, Arun Kumar wrote: Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 That looks like a problem that you should talk with Digium Support about. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TC400B load problem
Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 please help thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compile error
hi vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c this file and look for line that says 2.6.19 change it to 2.6.18 and save and compile arun On 5/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, May 04, 2007 at 01:55:20PM -0400, mail-lists wrote: I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Please test the patch in http://bugs.digium.com/view.php?id=9006 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Hi I already tried asterisk manager but Im not able to get status for each queue member. thanks On 5/8/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi, you can use an AGI to connect to asterisk manager and retrieve the info you need about the queue. Hope it helps Arun Kumar ha scritto: Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Queue Status
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] Change Codec
Here is some more details about my setup: Customer - PRI - Server A with G.729 - IAX Trunk G729 - Server B no G729 (pass through) - Snom Phone with G729 with incoming call there is no problem with when I try to make outbound and want to play some prompt on server b Im not able. in server B sip.conf : disallow=all allow=g729 if I'll write then Im able to make outbound call but not able to play any kind of prompt to the user who is making outbound call if disallow=all allow=ulaw and allow=g729 Im able to play some prompt but not able to make calls to the customer. thanks arun On 5/1/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Put similar allow/disallow statements in the sip or iax entry you create for your outbound ip calls. Be aware that if you use different codecs for phones and your termination provider, all media will have to go through asterisk and you will incur the processing overhead of codec conversion. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Arun Kumar *Sent:* Tuesday, May 01, 2007 9:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Change Codec Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Answer
Hi this is my setup: Customer - PRI - Server A with G729 - IAX2 Trunk(G729) - Server B - SIP Exten allowed codec=g729 - Snom phone Agents setup is working fine. I want when my agents are not available (queue) like not logged in or all are busy so no calls should come to my server b from server a I want my server a to not forward that call to my server b. Please guide me. Ive configured all my queue, sip exten on server b. server a is doing the routing of incoming calls to server b. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Status
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API Output
Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. ?php $strHost = 127.0.0.1; $strUser = cron; $strSecret = 1234; $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: Login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: QueueStatus\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); fclose($oSocket); ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Codec
Here is some more details about my setup: Customer - PRI - Server A with G.729 - IAX Trunk G729 - Server B no G729 (pass through) - Snom Phone with G729 with incoming call there is no problem with when I try to make outbound and want to play some prompt on server b Im not able. in server B sip.conf : disallow=all allow=g729 if I'll write then Im able to make outbound call but not able to play any kind of prompt to the user who is making outbound call if disallow=all allow=ulaw and allow=g729 Im able to play some prompt but not able to make calls to the customer. thanks arun On 5/1/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Put similar allow/disallow statements in the sip or iax entry you create for your outbound ip calls. Be aware that if you use different codecs for phones and your termination provider, all media will have to go through asterisk and you will incur the processing overhead of codec conversion. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Arun Kumar *Sent:* Tuesday, May 01, 2007 9:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Change Codec Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change Codec
Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] don't want call to get answered
In my * box I've configured two queues and incoming number and whenever any
one calls those number call comes to my *box and it sends call to my agents
in queue. but if no agent is available it still answer the call. Is there
any why when my agents are not available I don't want call to get answered.
Here is my dialplan:
exten = ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten = ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten = ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten = ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten = ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten = ,6,Goto(out-of-hours,5003,1)
exten = ,7,Answer()
exten = ,8,Playback(custom/next-avail-advisor)
exten =
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten = ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten = ,11,Queue(kbsupport,t)
exten = ,12,Hangup()
thanks
arun
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Re: [asterisk-users] Call Connection Problem
Hi, I'm using VoIP Service provider to place a call and I'm watching Asterisk CLI but it works fine but out of 5 tries it connects 1 time properly so there is no problem in placing the call b'coz I'm getting one call. thanks On 4/24/07, Nicholas Campion [EMAIL PROTECTED] wrote: To help me understand the problem, let me see if i have the environment straight. How are you connecting to the PSTN (to call your land line) FXO? VoIP Service Provider? How do you know Asterisk CLI is placing the call (are you watching the console?). If you are watching the console try and boost the debug / verbose settings and see if any extra information is provided. It sounds like (from your description) the script is working find from asterisk's point of view, but whatever sip/aix/whatever endpoing you are connecting to is failing to place the call to the land line. I'll need more information to help further. On 4/24/07, Arun Kumar [EMAIL PROTECTED] wrote: Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but I don't receive call on my land line and it starts playing the IVR. Please guide me how to solve the problem. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Connection Problem
Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but I don't receive call on my land line and it starts playing the IVR. Please guide me how to solve the problem. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exten Length
Hi, I've configured my exten.conf for few exten. But I'm curious to know how long can be my exten like (exten = XXX.). Is there any limit for this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my hard phone to make calls. when my exten length is 14 then calls goes immed. without any problem but when I change length from 14 to 15 call goes but when I dial 10 times I get only 1 or 2 connect (that is call never lends on my server if length is 15) but if I change length to 14 then 10/10 connects. any advice thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID Auth
Hi, in my dial plan I've configured two trunks to make outbound calls (one for national calls and other international). I want to allow only 2-3 extension to make use of my international trunk to make outbound calls so I want some kind of auth. based on their callerid . Please guide. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue Call Transfer
Hi I've configured the queue on my asterisk box and everything is working fine. In my queue I've 3 agents logged in the queue. When call comes they are able to receive the calls without any problem. But some time they are on break and there extension rings and no one is there to answer the call (we don't want them to log off from the queue) but we have one normal user in the same asterisk box registered so I want he dial some thing from his phone and that call should come to that normal user. Please advice. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ser as IVR
Hi, Is it possible to design an IVR using SER ? If yes please advice. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
I've installed zaptel on FreeBSD and when I try to load ztdummy module I get this error kldload: can't load ztdummy.ko No such file or directory. and when I do ztcfg:- Notice: Configuration file is /usr/local/etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' Keyword: [loadzone], Value: [us] Keyword: [defaultzone], Value: [us] 1 error(s) detected thanks On 4/17/07, Bryan M. Johns [EMAIL PROTECTED] wrote: Install zaptel and only enable the ztdummy module. As long as you are not running in a VM, this will supply you the timing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Arun Kumar [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Thomas Kenyon [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 4:54:47 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] No of Calls how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No of Calls
Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. thanks On 4/17/07, -- [ UxBoD ] -- [EMAIL PROTECTED] wrote: http://site.asteriskguide.com/bandcalc/bandcalc.php On Tue, 17 Apr 2007 11:54:28 +0400, Arun Kumar [EMAIL PROTECTED] wrote: Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card). thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adding Noise or background noise
Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding some noise /echo /latency or something. Please guide me. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Number of calls
HI, Here is my setup: USERS - PSTN - Service Provider - Asteriskbox1 - IAX2 trunk - Internet - IAX2 trunk - Asteriskbox2 -Sip Clients between asteriskbox1 and asterisk box2, I've VPN configured. from Asteriskbox2 to internet my line speed is 1MB. Is there any why that I can calculate how many number of concurrent calls I can place / receive. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] 11.2:5060..Call-ID: [EMAIL PROTECTED]: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4 ;received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED]:5060;tag=as78bcde29..Call-ID: [EMAIL PROTECTED] ..CSeq: 1 INVITE. .User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:800942@AsteriskIP..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;rece ived=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] :5060;tag=as78bcde29..Call-ID: [EMAIL PROTECTED] ..CSeq: 1 INVITE..User -Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip: 800942@AsteriskIP..Content-Type: application/sdp..Content-Length: 182v=0..o=root 2156 2156 IN IP4 AsteriskIP..s=session..c=IN IP4 Asterisk..t=0 0..m=audio 5676 RTP/AVP 18..a=rtpmap:18 G729/80 00..a=fmtp:18 annexb=no..a=silenceSupp:off - - - -.. # U PROVIDER-IP:5060 - AsteriskIP:5060 ACK sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..To: sip:[EMAIL PROTECTED]:5060;tag=as7 8bcde29..From: sip:PROVIDER-IP;tag=3380960452-790279..Contact: sip:PROVIDER-IP:5060..Call-ID: [EMAIL PROTECTED]: 1 ACK..Via: SIP/2.0/UDP PROVIDER-IP:5060; branch=z9hG4bK74ac10cb8c5d89375bf77d4aaa15fcea..Content-Length: 0 # U PROVIDER-IP:5060 - AsteriskIP:5060 BYE sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..To: sip:[EMAIL PROTECTED]:5060;tag=as7 8bcde29..From: sip:PROVIDER-IP;tag=3380960452-790279..Contact: sip:PROVIDER-IP:5060..Call-ID: [EMAIL PROTECTED]: 2 BYE..Via: SIP/2.0/UDP PROVIDER-IP:5060; branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27;rece ived=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED]:5060;tag=as78bcde29..Call-ID: [EMAIL PROTECTED]: 2 BYE..User-Ag ent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:8009 422419@AsteriskIP..Content-Length: 0 Any help appreciated Thanks! Rajeev On 2/20/07, Arun Kumar [EMAIL PROTECTED] wrote: Instead of forwarding to IAX softphone if I'll play some music same thing is happening in this case also. On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote: Without seeing your config files my guess would be that this is something to do with a bad codec negotiation. I'd bet that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP
[asterisk-users] Asterisk Inbound Problem
HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
Instead of forwarding to IAX softphone if I'll play some music same thing is happening in this case also. On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote: Without seeing your config files my guess would be that this is something to do with a bad codec negotiation. I'd bet that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Load Balancing
use LCR is really good. On 1/22/07, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me .. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Happy X-mas
Hi All, Wish you a very HAPPY and Merry Christmas to all and your beloved once. Arun On 12/23/06, Josué Conti [EMAIL PROTECTED] wrote: Hi ALL, ** I like very to desire you and your family, a Merry Christmas, with much love, peace, professional and personal success. Best Regards Josue 2006/12/23, raviprakash sunkara [EMAIL PROTECTED]: Hello * * * * * * * * * * * * * Happy X-mas and Adv Happy New Year ... ** * -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASterisk and SER
HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re:Call Transfers in SER + Asterisk
HI,
thanks for your reply. Here is my ser.cfg and other config files please
guide me.
ser.cfg
--
debug=5
fork=no
log_stderror=yes
listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE
port=5060
children=4
dns=no
rev_dns=no
fifo=/tmp/ser_fifo
fifo_db_url=mysql://ser:[EMAIL PROTECTED]/ser
loadmodule /usr/lib/ser/modules/mysql.so
loadmodule /usr/lib/ser/modules/sl.so
loadmodule /usr/lib/ser/modules/tm.so
loadmodule /usr/lib/ser/modules/rr.so
loadmodule /usr/lib/ser/modules/maxfwd.so
loadmodule /usr/lib/ser/modules/usrloc.so
loadmodule /usr/lib/ser/modules/registrar.so
loadmodule /usr/lib/ser/modules/auth.so
loadmodule /usr/lib/ser/modules/auth_db.so
loadmodule /usr/lib/ser/modules/uri.so
loadmodule /usr/lib/ser/modules/uri_db.so
loadmodule /usr/lib/ser/modules/domain.so
loadmodule /usr/lib/ser/modules/mediaproxy.so
loadmodule /usr/lib/ser/modules/nathelper.so
loadmodule /usr/lib/ser/modules/textops.so
loadmodule /usr/lib/ser/modules/avpops.so
loadmodule /usr/lib/ser/modules/permissions.so
modparam(auth_db|permissions|uri_db|usrloc|domain, db_url,
mysql://ser:[EMAIL PROTECTED]/ser)
modparam(auth_db, calculate_ha1, 1)
modparam(auth_db, password_column, password)
modparam(nathelper, rtpproxy_disable, 1)
modparam(nathelper, natping_interval, 0)
modparam(mediaproxy,natping_interval, 30)
modparam(mediaproxy,mediaproxy_socket, /var/run/mediaproxy.sock)
modparam(mediaproxy,sip_asymmetrics,/etc/ser/sip-clients)
modparam(mediaproxy,rtp_asymmetrics,/etc/ser/rtp-clients)
modparam(usrloc, db_mode, 2)
modparam(registrar, nat_flag, 6)
modparam(rr, enable_full_lr, 1)
modparam(tm, fr_inv_timer, 27)
modparam(tm, fr_inv_timer_avp, inv_timeout)
modparam(permissions, db_mode, 1)
modparam(permissions, trusted_table, trusted)
# - request routing logic ---
# main routing logic
route {
# -
# Sanity Check Section
# -
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
break;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
break;
};
# -
# Record Route Section
# -
if (method==INVITE client_nat_test(3)) {
# INSERT PROXY IP ADDRESS HERE
record_route_preset( 2xx.xxx.xxx.xxx:5060;nat=yes);
} else if (method!=REGISTER) {
record_route();
};
# -
# Call Tear Down Section
# -
if (method==BYE || method==CANCEL) {
end_media_session();
};
# -
# Loose Route Section
# -
if (loose_route()) {
if ((method==INVITE || method==REFER) !has_totag())
{
sl_send_reply(403, Use From=ID);
break;
};
if (method==INVITE)
{
if (!allow_trusted())
{
if (!proxy_authorize(,subscriber))
{
proxy_challenge(,0);
break;
} else if (!check_from()) {
sl_send_reply(403, user From=ID);
break;
};
consume_credentials();
};
if (client_nat_test(3) || search(^Route:.*;nat=yes)){
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
# -
# Call Type Processing Section
# -
if (!is_uri_host_local()) {
if (is_from_local() || allow_trusted()) {
route(4);
route(1);
} else { sl_send_reply(403, Forbidden-two);
};
break;
};
if (method==ACK) {
route(1);
break;
} if (method==CANCEL) {
route(1);
break;
} else if (method==INVITE) {
route(3);
break;
}
[asterisk-users] Asterisk with SER
HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-20x
hican you please post some user or config guide.thanks in advancearunOn 10/24/06, Ed Greenberg [EMAIL PROTECTED] wrote:I will sign in with good experiences with MP124 and Mediant 1000. I have an MP202 under test.--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas[EMAIL PROTECTED] wrote: I have used AudioCodes MP 102, 104 and 108, both FXS and FXO. I have also used AudioCodes Mediant 2000. I can tell you that these are good devices. There are also many other media gateways that have a lot of facilities, but many of these implement those facilities in software. AudioCodes has also a quite good – let's say -- hardware support. I haven't used MP20x. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Andrew Joakimsen Sent: Monday, October 23, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-20x Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them...___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iax2 show netstat
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/callaus-3 265 -10-1 -1 0 -1 0
0 40 0 0 00 0
IAX2/2025-4 5 -10-1 -1 0 -1 10
17 92 5 0 10 10
IAX2/callaus-71000 -10-1 -1 0 -1 0
00 0 0 00 0
IAX2/2002-15 4 -10-1 -1 0 -1 12
17 75 3 0 00 11
4 active IAX channels
thanks
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[asterisk-users] Iax Netstat Output
can please some one tell me where is what wrong.iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO KpktsIAX2/callaus-3 265 -1 0 -1 -1 0 -1 00 40 0 0 0 0 0IAX2/2025-4 5 -1 0 -1 -1 0 -1 10 17 92 5 0 1 0 10IAX2/callaus-7 1000 -1 0 -1 -1 0 -1 00 0 0 0 0 0 0IAX2/2002-15 4 -1 0 -1 -1 0 -1 12 17 75 3 0 0 0 114 active IAX channelsthanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
b'coz I have same setup at other client is working fine no problem.On 9/22/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote: can please some one tell me where is what wrong. iax2 show netstats LOCAL - REMOTE ChannelRTTJitDelLost %DropOOOKpkts JitDelLost %DropOOOKpkts IAX2/callaus-3 265 -10-1-1 0 -10 0 40 0 0 000 IAX2/2025-45 -10-1-1 0 -1 10 17 92 5 0 10 10 IAX2/callaus-71000 -10-1-1 0 -10 00 0 0 000 IAX2/2002-15 4 -10-1-1 0 -1 12 17 75 3 0 00 11 4 active IAX channelsCould you please tell us why do you believe that there is actuallysomething wrong?Or is this a certain Asterisk-competence quiz that I have just failed? --Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755iax:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iax Netstat Output
HiI've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem.here is the iax netstat output Channel RTT Jit Del Lost Drop OOO Kpkts Jit Del Lost Drop OOO Kpkts 3 Traffic from Server to Agent 4 IAX2/2003-29 6 -1 0 -1 0 -1 228 18 89 1294 0 127 226 5 IAX2/2006-18 5 -1 0 -1 0 -1 83 18 87 934 0 1 826 IAX2/2021-11 11 -1 0 -1 0 -1 30 20 77 158 0 1 28 7 IAX2/2021-12 9 -1 0 -1 0 -1 45 20 80 167 0 0 448 IAX2/2021-31 8 -1 0 -1 0 -1 74 18 91 429 0 4 73 9 IAX2/2022-13 6 -1 0 -1 0 -1 123 17 94 740 0 1 12110 IAX2/2023-6 11 -1 0 -1 0 -1 229 19 97 2114 0 46 226 11 IAX2/2024-49 9 -1 0 -1 0 -1 45 18 76 202 0 1 4412 13 Traffic from Server to Minutes Provider 14 IAX2/callaus-15 1000 -1 0 -1 0 -1 0 0 0 0 0 0 0 15 IAX2/callaus-30 1000 -1 0 -1 0 -1 0 0 0 0 0 0 016 IAX2/callaus-34 259 -1 0 -1 0 -1 5 0 40 0 0 0 0 17 IAX2/callaus-4 502 -1 0 -1 0 -1 1 0 40 0 0 0 018 IAX2/callaus-40 260 -1 0 -1 0 -1 2 0 40 0 0 0 0 19 IAX2/callaus-5 1000 -1 0 -1 0 -1 0 0 0 0 0 0 020 IAX2/callaus-7 259 -1 0 -1 0 -1 1 0 40 0 0 0 0 21 IAX2/velilevox-19 256 -1 0 -1 0 -1 14 0 40 0 0 0 0thankarun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
no zap - iax2 - iax2 only iax2 - iax2 - iax2thanksOn 9/21/06, Ma Zhiyong [EMAIL PROTECTED] wrote:I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Outgoing Spool Failed
hithanks for reply.I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ?thanks arunOn 9/8/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Arun Kumar wrote: hi my asterisk -r shows me Most of the times Outgoing Spool Failed. Can some one tell me why is it happening and how to solve this issue. Is it a problem ?You'd need to provide more information.Does it work when you call normally?Are you spooling lots of calls at the same time?Show us your spool file.- -- Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone)http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)-BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFFAHZFS6d5vy0jeVcRAm6cAJ9dCQsEPPs7HWRk/hCcVjNVBSiaTwCfaZDoAafoRpj4XhD8LoMvXkgAlSc= =7bSO-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Outgoing Spool Failed
himy asterisk -r shows me Most of the times Outgoing Spool Failed. Can some one tell me why is it happening and how to solve this issue. Is it a problem ? thanks in advance.arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + centos 4.3
hi,can you describe what you want.../ArunOn 7/14/06, varun [EMAIL PROTECTED] wrote:Hello,We were able to get asterisk going withX100p cards on centos 4.2.But could on centos 4.3 due to kernelissues.Anybody has faced this issue ?And how do sort it out so that wecan use centos 4.3 ?ThanksVarun___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed
Hi,Here are the step by step instructions for setting up a brand new AudiocodesFXS gateway for use with an Asterisk server: Connect the gateway to a network switch and connect a computer to the same switch. Then configure the IP address of the computer to 10.1.10.2. Then runyour web browser and point it to http://10.1.10.10 and login using the information below.Default IP address: 10.1.10.10Default user name: AdminDefault password: AdminGoto Quick Setup and change the following:IP Address = Set to the new IP address of the AudioCodes gateway Subnet Mask = Set to the correct netmask for your local network Default Gateway Address = Set to the correct gateway IP address for yourlocal network Working With Proxy = Set to Yes Proxy IP Address = Set to the IP address of the Asterisk server Enable Registration = Set to EnableRestart the gateway then log back in using the new IP address.Goto Protocol Management - Protocol Definition - Proxy Registration Registrar IP Address = Set to the IP address of the Asterisk server Registration Time = Set to 60Subscription Mode = Set to Per EndpointAuthentication Mode = Set to Per Endpoint Goto Protocol Management - Protocol Definition - DTMF Dialing Max Digits In Phone Num = Set to a large enough number such as 32Goto Protocol Management - Protocol Definition - Coders Add coders as needed You need to set at least G.711U-lawGoto Protocol Management - Endpoint Settings - Authentication Set SIP username and password for each portGoto Protocol Management - Endpoint Phone Numbers Enter an extension (phone) number for every used channelYour AudioCodes gateway is now ready.../Arun[EMAIL PROTECTED] www.intelegentnetworks.comOn 6/27/06, Mark Adams [EMAIL PROTECTED] wrote: Hello, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special configuration nor am I going to use it for any incoming calls. I would like to just have the gateways register and properly send calls out and relay DTMF tones. After I get them up and running I should have the manual read and digested by then and I will be good to go. Anyone interested please email me off list Mark Adams Infinity Marketing 216-334-9304 ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk auto-dial Help
Hi,When you originate a call asterisk essentially callouts to the Specified channel and the when answers connects the the context,extension,priority. What if I want my dial plan to make the origination call and the destination call. What would I specify for my dialplan/callout file?thanks in advance../Arun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a2billing
Hi,you check your a2billing.conf file:; Please enter here the file you want to play when we prompt the calling party to enter his destination number; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011 file_conf_enter_destination = prepaid-enter-destI think this file should help you.../Arun[EMAIL PROTECTED] www.intelegentnetworks.comOn 6/28/06, Khaled Chehab [EMAIL PROTECTED] wrote: I am using a2billing as billing system on tixbox but I have a problem since the user call the destination number ,the ivr tell him about him amount and ask him to enter the destination number ,my question is how can I let the user call the destination directly . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free sun boxes
Hi, What is the Location. I'm studying in India. Is it possible. thanks ArunOn 6/17/06, Bob Knight [EMAIL PROTECTED] wrote: I have 4 sparc based sun boxes I am about to pay money so I canget rid of them.They are running older versions of Solaris.You should be able to load Solaris 10 and play around with *on them.Time to clean the office: 3 Ultra 51 Sparcstation 5I also have a box full of Sun keyboards and mice.Contact me offline if you want them.I've had many good years of development on them and it killsme to just toss them, but the office is just too damn cluttered. thanks, bk...--Bob Knight[-w] the work option[EMAIL PROTECTED]925-449-9163___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft CRM Asterisk
Hi Calvis, Its good if I can help you in any why with this project. thanks ../ArunOn 6/2/06, calvis [EMAIL PROTECTED] wrote: Has anyone done any integration with Asterisk Microsoft Dynamics CRM?Ijust wanted to check with the list before I pursue a project with the aboveintegration.In addition, if anyone would be interested in such an integration let me know, and I will keep you posted on the results.Thanks,Charles AlvisInternet Technology Group, Inc.Redmond,WA___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help Needed
Can you please send me a link for getting the details (like features, prices etc) for this card? Thanks Regards Arun -Original Message- From: Lubomir Christov [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 20:58 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help Needed Hello, you will need PhoneJACK (PCI or ISA) Few days ago they announced that there is a new PhoneJACK PCI available - with new DSP and etc. Best regards Lubo Arun Kumar Sharma, Noida wrote: Thanks Adam, This document provides me a high level architecture of Asterisk. Can you please tell me if I want to evaluate Asterisk on an Intel PC which Quicknet hardware will be required to just run a POTS to SIP call? Thank you once again for very fast response. Regards Arun -Original Message- From: Low, Adam [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 19:11 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] [Asterisk-Users]Help Needed http://www.digium.com/handbook-draft.pdf -Original Message- From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 15:27 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [Asterisk-Users]Help Needed * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault with chan_oh323
Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun -Original Message- From: Mark Thompson [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault with chan_oh323 This also happened to me when I was using the same codec with both oh323 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection worked. I also tried h323 instead of oh323 which works okay but you have to use earlier versions of pwlib and openh323. Mark -Original Message- From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] Sent: 16 July 2003 23:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault with chan_oh323 Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it Trying and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=phone number. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help Needed
Thanks Adam, This document provides me a high level architecture of Asterisk. Can you please tell me if I want to evaluate Asterisk on an Intel PC which Quicknet hardware will be required to just run a POTS to SIP call? Thank you once again for very fast response. Regards Arun -Original Message- From: Low, Adam [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 19:11 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] [Asterisk-Users]Help Needed http://www.digium.com/handbook-draft.pdf -Original Message- From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 15:27 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [Asterisk-Users]Help Needed * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
