[asterisk-users] Needed changes in Asterisk to change the SIP port to 5062
Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observed on my server console that my server is registered with our VoIP provider with 5062 port. But, I am unable to make outgoing calls. Do I need to modify any other settings in Asterisk? Look forward to your response. Thank you. Regards, Chandra. - Need a vacation? Get great deals to amazing places on Yahoo! Travel. ___ --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] SugarCRM Integration
Yes. You are right. You can integrate Sugar with Trixbox very easily. You can customize it also. Thanks, Chandra Joseph Bajin [EMAIL PROTECTED] wrote: I'd like to know as well about this. On 6/1/07, Diego Quintana Cruz wrote: Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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 - Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.___ --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+IMATE PDAL Configuration+Softphone
Hi friends, I am planning to buy IMate PDAL mobile phone. This phone supports Wi-Fi. So, I want to configure my Asterisk extension with this mobile. For this, I think that need to install a softphone. Can anybody tell me the softphone that is compatible with this mobile and configuration settings? Look forward to your response. Thank you. Regards, Chandra. - Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA.___ --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] Wi-Fi+Wireless Router
Hi Friends, I am planning to buy IMate PDAL mobile phone. This contains Wi-Fi 802.11b/g feature. So, Is it possible to get internet using my wireless router in my office? Look forward to your response. Thank you. Regards, Chandra. - Got a little couch potato? Check out fun summer activities for kids.___ --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] Blackberry 8800+VoIP Configuration
Hi Friends, I bought a new Black Berry 8800 mobile. This mobile is coming with Wi-Fi feature. After turning on this Wi-Fi feature in my mobile, It is not detecting my wireless router in our office. How can I do this? How can I configure my VoIP (SIP) server in this Blackberry 8800 mobile? I tried a lot to do the above things in my mobile. But, I failed. Look forward to your response. Thank you. Regards, Chandra. - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --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] SIP Echo
Hi, Did you implement QoS (Quality of Service) in your network? Thanks. Regards, Chandra Stephen Bosch [EMAIL PROTECTED] wrote: Asterisk wrote: I tried with the ping ... all of the phones respond in ca. 0.3ms, so network seems to be OK. More than 90% of CPU on * box is idle even in peak times, so this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, but did anyone ever experience any handset problems with Polycom IP SoundPoint 430 :-) ? Headsets are a terrible source of echo. Are you using a headset amplifier? Polycom specifically recommends use of an amplifier with the SoundPoint IP phones (most of the newer ones have integrated echo cancellation). -Stephen- ___ --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 - Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___ --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] Please help me finding good A-Z provider
No, I had very very bad experience with Teliax. I recommend Inphonex (http://www.inphonex.com/). Thanks, Chandra. Anthony Francis [EMAIL PROTECTED] wrote: teliax? teliax.com VoIP User wrote: Hi Everyone, can you please recommend me a good VoIP provider as I am not satisfied by my current provider. Does not matter what protocol it uses. I'm looking for good rates, stable quality and not so big prepayment required. Thanks to all. ___ --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 - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ --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] Any other softPBX like Asterisk?
Thank you. I will go through these softwares. Luca Corti [EMAIL PROTECTED] wrote: On Fri, 2007-05-11 at 07:33 -0300, Roberto Pereyra wrote: - OpenPBX - Freeswitch Other: sipX Yet another: Yate http://yate.null.ro/pmwiki/ ciao Luca ___ --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 - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --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] Any other softPBX like Asterisk?
Hi Friends, Can anybody tell me other softPBX softwares like Asterisk? Thank you. Regards, Chandra. - Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta.___ --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] My Polycom IP 501 is formatted its file systemitself
Hi Steven, Thank you for very much for your response. I solved this problem. I found these files in Internet and setup FTP server and uploaded these files into that. Now, my phone is working. Thank you. Regards, Chandra. Steven Ringwald [EMAIL PROTECTED] wrote: No problem. Sorry about the delay. Unfortunately, I no longer work for a Polycom reseller, so I can't give you a simple link to the files that you need (someone else on the list might be able to help you out off-list with this). Sometimes, you can find them online by searching google with Polycom 501 firmware. Sorry that I can't be of more help. If I come across the image files, I will send you an email and let you know. Best of luck! Steve Crazy Boy wrote: Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. */Steve Totaro /* wrote: We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file system itself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - Noah ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file systemitself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. Steve Totaro [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, April 25, 2007 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file systemitself Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - Noah If it just keeps rebooting with an error about not being able to load the application or something like that, I had one do that. I found a fix via google. If this is the issue, post back and I will see if I can find the link. It is an easy fix. Thanks, Steve ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file system itself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. »Steven Ringwald« [EMAIL PROTECTED] wrote: Noah Miller wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. If you can get to the boot menu (where it offers to let you configure a server to boot off of), you can recover with a firmware image. You usually can get these only from resellers (because Polycom doesn't want to deal with customer support on an individual basis). Let me know if you can get this far; I might be able to help. Steve ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Trixbox/FreePBX
Hi, Write down your problem clearly. Thanks [EMAIL PROTECTED] wrote: Hello, Installed Trixbox with a digium card and it is taking 2 rings for it to pick up. Any suggestions how to have the system pickup immediately? Thanks, Neal ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] How to check my voice mail from outside landline?
Hi Adam and Noah, Thank you for your patience and response. I will try to do it now. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Adam - there must be an easier way, but since i only have asterisk and a couple of ATAs (spa 3k), i've set one up to give a dial tone to the incoming caller on the FXO port. This way, dialing a pstn number i get another dial tone to access internal extensions, such as voicemail. That's a perfectly good way to do it. I don't know what your primary incoming line does, but if it goes to a queue or a message, you could also have a special DTMF key to break out and go to VoicemailMain or whatever. That would save you from having to have a dedicated line for outside access to your extensions. Or, if you have a live person answering your main number, and you don't want to bother him/her, you could get a really cheap VoIP DID and dedicate it for access to your extensions. - Noah ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Polycom IP 501 is displaying wrong time
Hi Steve,
Thank you for your help and information. You told me that you found another
one. Can you tell me that another one please?
Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:v\:*
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:*
{behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) } You can use the web interface
and set it to -5 gmt. Google for free NTP servers. I used to use
time.nist.gov and got mixed results. I found another one that works almost all
of the time.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Thursday, April 19, 2007 7:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
display screen. How can I set the New York time? What value I have to give
to GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
-
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Hello Chris, Thank you very much for your help. I am getting time now. Regards, Chandra. Chris Mason (Lists) [EMAIL PROTECTED] wrote: If your phone is getting its parameters by DHCP from a linux server, add the NTP server option to that server: in /etc/dhcpd.conf option time-servers 192.168.0.3; If your phone is getting an NTP server setting by DHCP server, you can't override that from any setting. I came across this where a polycome 501 was connected to the internet directly and comcast was setting NTP to 10.10.x.x, which was ridiculous. Their tech support could never understand why this was a problem and would not address the problem despite repeated calls. If there is no setting for NTP in your DHCP server, for example if you have a linksys router for your network, you can set the DHCP server in sip.cfg and set the offset in seconds. The offset for AST is -14400 for example. If you are configuring by web browser, you can set the timezone and ntp server that way. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Polycom IP 501 is displaying wrong time
Hi Bruno, Thank you very much for your needful help. I am getting time now. Regards, Chandra. Bruno De Luca [EMAIL PROTECTED] wrote:Hi, this code is for italian time is inside the sip.cfg file. SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=192.168.0.8 tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=3600 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=1 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ Bruno. Dave Miller wrote: Crazy Boy wrote on 4/19/07 11:41 PM: Thank you for your response. As you said, I set it for -5. But, its displaying wrong time. I don't enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Yeah, there's no way to set the clock except by using an NTP server, so you need to set one. ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] How to check my voice mail from outside landline?
Hi Friends, I installed and configured Asterisk. I am getting my voice mail to my email as attachments. Well. We can check our voice mail by dialing *98. But, I want to check my voice mails by dialing our DID number from a outside telephone. How can I do this? Please help me. Look forward to your response. Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file system itself
Hi Noah, Thank you for your response. Yes, It is giving boot menu and giving a chance to configure boot server. What can I do now? Please tell me. Thank you. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - Noah ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file systemitself
Hi Steve, Thank you for your response. Yes, It is giving boot menu and giving a chance to configure boot server. What can I do now? Please tell me. Thank you. Regards, Chandra. Steve Totaro [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, April 25, 2007 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file systemitself Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - Noah If it just keeps rebooting with an error about not being able to load the application or something like that, I had one do that. I found a fix via google. If this is the issue, post back and I will see if I can find the link. It is an easy fix. Thanks, Steve ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] My Polycom IP 501 is formatted its file system itself
Hi, We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. Look forward to your response. Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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 config for Apple IPhone
Hi Friends, I want to buy Apple IPhone mobile. How to configure my Asterisk server in this mobile? Is this mobile supports VoIP configuration? Look forward to your response. Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Apple IPhone mobile is released in India?
Hi Friends, Is Apple IPhone mobile is released in India? Is Apple IPhone mobile is released in USA? If IPhone is released in India, Can you tell me any Apple authorized showroom in Hyderabad (Andhrapradesh, India)? Look forward to your response. Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Polycom IP 501 is displaying wrong time
Hi,
Thank you for your response. As you said, I set it for -5. But, its displaying
wrong time. I don't enter any SNTP Server. Is it must? How can I solve this
problem? Can you tell me?
Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:v\:*
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:*
{behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) } You can use the web interface
and set it to -5 gmt. Google for free NTP servers. I used to use
time.nist.gov and got mixed results. I found another one that works almost all
of the time.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Thursday, April 19, 2007 7:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
display screen. How can I set the New York time? What value I have to give
to GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi Noah, Thank you for your response. As you said, I tried to enter -18000 in GMT offset field. But, its not taking input from the phone dial pad or key board. Its giving chance to select the value from -12 to 12. I dont enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Thank you. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra - This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the New York time? What value I have to give to GMT offset value? The GMT offset value is in seconds. So, for example, the value to use for EST is -18000, because EST is -5 hours from GMT (-5 x 3600 = -18000). - Noah ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Polycom IP 501 is displaying wrong time
Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the New York time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] BSNL caller ID (India)
Hello Mr. Sanjay, I tried a lot to get caller ID in India. But, It doesn't work. I came to know that Its not possible to get caller ID in India (Not only in India, don't get caller ID in some countrys). Thank you. Regards, Chandra. Sanjay Rajdev [EMAIL PROTECTED] wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] How can i send voicemail to multiple email IDs?
Hi, I have created two extensions (156157) with voicemail enabled. When I receive a call from outside, my IVR is responded. When user press 156, if he (156) unable to answer the phone, the voice mail will be goes to 156 and 157 email IDs. I mean, I want to send voice mail to multiple email addresses. How can i do this? Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] How can i add multiple callerids to an inbound route?
Hi, I have configured the below things: Extensions Trunk Outbound route Inbound route IVR Ring group If anybody call to my DID number, my IVR is responded. After that, if he press 1, then Ring group will be activated. All are working fine. My Problem: I want to avoid IVR for some phone numbers depends on their called IDs. If my common users will call to my DID number, my ring group will be activated directly without playing IVR. We will do it by adding one by one calledIDs in inbound route and redirect it to ring group. This solution is suitable for 5 or 10 caller IDs. But, I have 200 standard caller IDs. Its difficult to add all these one by one. Is there any script or any other way to handle to do this at a time? Thank you. Regards, Chandra. - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] How can i send voicemail to multiple email IDs?
Hi Dawson, Thank you for your response. I hope this is the good solution as said by you. Regards, Chandra. [EMAIL PROTECTED] wrote: Hi Chandra, One option is to set up a group on your mail server containing multiple addresses and use the group name as the email address within Asterisk. rgds, Phil. Crazy Boy oo.comTo Sent by: [EMAIL PROTECTED] asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [asterisk-users] How can i send voicemail to multiple email IDs? 13/04/2007 09:37 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion ists.digium.com Hi, I have created two extensions (156157) with voicemail enabled. When I receive a call from outside, my IVR is responded. When user press 156, if he (156) unable to answer the phone, the voice mail will be goes to 156 and 157 email IDs. I mean, I want to send voice mail to multiple email addresses. How can i do this? Thank you. Regards, Chandra. Ahhh...imagining that irresistible new car smell? Check out new cars at Yahoo! Autos. ___ --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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Anybody using VoiceRD or Thinkbright service?
Hi Friends, I am planning to buy VoiceRD software to settingup my call centre and planning to use Thinkbright as VoIP provider. Anybody using the above one or two? If you are using any of the above, please tell me your opinions. Looking forward to your response. Thank you. Regards, Chandra. - Access over 1 million songs - Yahoo! Music Unlimited.___ --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] Polycom IP 501+India
Hi Friends, This is Chandra from India. I have installed and configured Asterisk in our company. I want to provide Polycom IP 501 model phones to our employees. I am unable to find the dealer for these phones in India. Where can I buy these phones in India? If anybody knows, please tell me the dealer address or phone number. This is very urgent. Looking forward to your response. Thank you. Regards, Chandra. - Everyone is raving about the all-new Yahoo! Mail beta.___ --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] Attn: DISA Experts(Strange problem with DISA)
Hi Friends,
I am facing a strange problem with DISA. I have installed and configured
Trixbox. I've created a secret extension i.e., 555 and called this extension in
Digital Receptionist using custom extension i.e., created in
extensions_custom.conf file.
When I call from my mobile phone to my PSTN number, which is connected to FXO
port, my IVR is responding. After entering my DISA extension(555), its asking
password. After entering correct password, its giving ringtone. Upto this, no
problem. The problem is coming here only. When I enter a USA number, its taking
the first digit of USA number twice.
For eg: If I enter 17187773456, its taking as: 117187773456
If I enter 917187773456, its taking as: 9917187773456
Its taking my input USA number correctly for sometimes and call is connected to
my mobile. I tried by changing the value of relaxdtmf from yes to no and vice
versa.
Here I am sending my config files. Please tell me the solution.
Extenstions_custom.conf contents:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _9.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN:1},90,tr)
exten = _9.,2,Hangup
Zapata.conf file contents:
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300
usecallerid=yes
relaxdtmf=yes
dtmfmode=rfc2833
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6
faxdetect=incoming
#include zapata-auto.conf
group=1
#include zapata_additional.conf
Please tell me the solution. Looking forward to your response. Thank you.
Regards,
Chandra.
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[asterisk-users] Asterisk is taking the first digit of my entered number twice. Why?
Hi Friends, I am working on DISA. When I call to my fxo number, its asking extension. I entered my secret DISA extension and its asking the PIN number. After that Asterisk is giving dial tone to dial a USA number. I am facing problem here only. When I entered a USA number, Asterisk is taking the first digit of my entered number twice. For eg: If I enter 17187773456, its taking as: 117187773456 If I enter 3247312653, its taking as: 33247312653 Here I am sending my zapata.conf file contents: [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 usecallerid=yes relaxdtmf=yes dtmfmode=rfc2833 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=6 faxdetect=incoming group=1 Please tell me the problem. Am I need to add the anything to this file? Looking forward to your response. Thank you. Regards, Chandra. - Everyone is raving about the all-new Yahoo! Mail beta.___ --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] Request for help with DISA (Not taking my input number correctly?)
Hi Steve,
Thank you for your response. As you said, i tried. But, no result. Here I am
sending my configuration file.
Contents in Zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300
usecallerid=yes
relaxdtmf=yes
dtmfmode=rfc2833
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6
group=1
Please tell me, if there are any modifications in config. files. So that, i
will test it again. Looking forward to your response. Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote: Crazy Boy wrote:
Hi,
Thank you for response. I configured DISA and its working sometimes
and not working sometimes. Here I am sending the configuration and
output on Asterisk server console:
Extensions.conf file content:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _X.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)
Output on server console:
-- Playing 'custom/v1' (language 'en')
== CDR updated on Zap/1-1
-- Executing Goto(Zap/1-1, custom-CLID|s|1) in new stack
-- Goto (custom-CLID,s,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing Authenticate(Zap/1-1, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing DISA(Zap/1-1, no-password|disa-ext) in new stack
-- Executing Dial(Zap/1-1,
IAX2/[EMAIL PROTECTED]/187773456|30|tr) in new stack
-- Called [EMAIL PROTECTED]/187773456
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/1-1'
What is happening is:
1) I called my zap number from my mobile
2) My IVR is responding
3) I entered a extension number to access DISA
4) Asterisk asked the secret (PIN) code to access DISA
5) I entered password of DISA
6) After validating the password, its giving Dial tone to dial a USA
number
7) I entered 17187773456 (This is a toll free number) to test
8) Call is going sometimes and call is not going sometimes. If we
observe on server console, its not taking my input number properly and
taking my input phone number wrongly.
9) I tested from other mobiles also. But, its not taking my input
number as i entered sometimes.
9) What is the wrong?
Please tell me. Looking forward to your response. Thank you.
Regards,
Chandra.
Do you have relaxdtmf set in your zap conf file? Try from a landline
phone and see if you have the same issue. If you have relaxdtmf=yes try
no and test again, or do the opposite. It is obvious that asterisk is
getting dtmf but it is mixing it up.
Also try dialing other companies IVRs and navigating the menus, maybe
your cell phone is just screwed up?
Is the call going through any other boxes that may have DTMF settings
misconfigured? I have seen DTMF come out in doubles (ie you press 911
and asterisk sees 99111).
Thanks,
Steve Totaro
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[asterisk-users] Request for help with DISA (Not taking my input number correctly?)
Hi,
Thank you for response. I configured DISA and its working sometimes and not
working sometimes. Here I am sending the configuration and output on Asterisk
server console:
Extensions.conf file content:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _X.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)
Output on server console:
-- Playing 'custom/v1' (language 'en')
== CDR updated on Zap/1-1
-- Executing Goto(Zap/1-1, custom-CLID|s|1) in new stack
-- Goto (custom-CLID,s,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing Authenticate(Zap/1-1, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing DISA(Zap/1-1, no-password|disa-ext) in new stack
-- Executing Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/187773456|30|tr) in
new stack
-- Called [EMAIL PROTECTED]/187773456
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/1-1'
What is happening is:
1) I called my zap number from my mobile
2) My IVR is responding
3) I entered a extension number to access DISA
4) Asterisk asked the secret (PIN) code to access DISA
5) I entered password of DISA
6) After validating the password, its giving Dial tone to dial a USA number
7) I entered 17187773456 (This is a toll free number) to test
8) Call is going sometimes and call is not going sometimes. If we observe on
server console, its not taking my input number properly and taking my input
phone number wrongly.
9) I tested from other mobiles also. But, its not taking my input number as i
entered sometimes.
9) What is the wrong?
Please tell me. Looking forward to your response. Thank you.
Regards,
Chandra.
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Re: [asterisk-users] Request for working config for DISA
Hi,
Thank you for your response. As you said, I have tested. But, its not going and
simply hangup. What I have to do? Please tell me. Thank you.
Regards,
Chandra.
zero massive [EMAIL PROTECTED] wrote: Here you go:
[Custom-CLID]
exten = s,1,Answer
exten = s,2,Authenticate(12345)
exten = s,15,Playback(after-the-tone)
exten = s,16,Playback(pls-entr-num-uwish2-call)
exten = s,18,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = s,19,Monitor(wav,${CALLFILENAME},m)
exten = s,20,DISA(no-password|from-internal|${CLIDArea})
On 11/22/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi Friends,
I have configured DISA. But, its not working. When I dial to my zap channel,
its asking to enter pin number. After entering PIN number, its giving
continuous engage sound and hangup. Can anybody send me correct working
configuration for DISA? Looking forward to your response. Thank you.
Regards,
Chandra.
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[asterisk-users] Request for working config for DISA
Hi Friends, I have configured DISA. But, its not working. When I dial to my zap channel, its asking to enter pin number. After entering PIN number, its giving continuous engage sound and hangup. Can anybody send me correct working configuration for DISA? Looking forward to your response. Thank you. Regards, Chandra. - Sponsored Link Get an Online or Campus degree - Associate's, Bachelor's, or Master's -in less than one year.___ --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] Attn:Peter, Gsalas, Tim-Help me to configure my NOKIA E70 Mobile with my Asterisk server
Hi Friends, Thank you for your response. Yesterday only, I configured my Nokia E70 mobile and its working fine. For group members convenience, here I am giving the configuration: Configuring the Nokia E70: Go to Menu - Tools - Settings - Connection - Sip Settings - Profile name: Olivetalk Service Profile: IETF Default Access Point: Olive Public user name: sip:[EMAIL PROTECTED] Use Compression: No Registration: Always On Use Security: no Proxy server settings: Proxy server address: sip:202.xxx.xxx.xxx Realm: asterisk User name: 102 Password: chandra Allow loose routing: Yes Transport Type: UDP Port: 5060 Registrar Server Settings: Registrar serv. addr.: sip:202.xxx.xxx.xxx Realm: asterisk User Name: 102 Password: chandra Transport Type: UDP Port: 5060 Go to Menu - Tools - Settings - Connection Internet tel. settings - Name : Olivetalk SIP profiles : Olivetalk I hope this information will be useful for remaining users. Thank you. Regards, Chandra. - Sponsored Link Mortgage rates near 39yr lows. $420,000 Mortgage for $1,399/mo - Calculate new house payment___ --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] How to use Voipjet or any Voip provider Trunk from my mobile through fxo and fxs ports?
Hi Friends,I have installed Asterisk and configured successfully. Now, I got a doubt. Here I am giving my configuration.1) 1 PSTN line connected to FXO port and created inbound route. (Ph. No: 233534)2) 1 Analog phone connected to FXS port and created ZAP extension with No. 1033) Configured "Voipjet" trunk to make international calls.All the above are working fine.Now, My problem is: I have to make international calls from my mobile through Voipjet trunk using my Asterisk server.When I make a call to 233534 from my mobile, call will automatically goes to 103. Its working fine. Now, I have to dial a international number (For eg: 1 718 777 3456) and call should be go through Voipjet trunk. How can I do this? Please tell me or suggest me a good link to do this.Looking forward to your response. Thank you.Regards,Chandra. Everyone is raving about the all-new Yahoo! Mail beta.___ --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] Help for registration with sipdiscount
Hi Friends,I have an account with sipdiscount.com. I configured my Asterisk server. When I try to make a call, its telling that "All circuits are busy". I tried in many ways. Can anybody send me correct working configuration for sipdiscount? Thanks in advance.Regards,Chandra. Check out the New Yahoo! Mail - Fire up a more powerful email and get things done faster. ___ --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] Is it possible to connect two servers using SIP?
Hi Friends,I have created SIP extensions in our two Asterisk servers. Now, I want to connect these two servers using SIP. I searched a lot in internet about this. But, I found that there is a possibility to connect two servers using IAX2 only. Is it possible to connect two Asterisk servers using SIP? If so, can you give me a tutorial link about this?Looking forward to your response. Thank you.Regards,Chandra. We have the perfect Group for you. Check out the handy changes to Yahoo! Groups. ___ --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 is not coming through sipgate.co.uk+Asterisk
Hi,I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100. I configured my Asterisk server with 0207100. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I made a call, Asterisk console is also not showing anything. But, sipgate website is showing my calls list. I thought that When I made a call from outside to my number, call is going to sipgate.co.uk and its not routing to my server. When I execute "sip show registry", its not displaying anything. Here I am giving my configuration details:My sip.conf file contents:[general]port = 5060bindaddr =
0.0.0.0qualify=nodisable=allallow=alawallow=alawallow=ulawallow=g729allow=gsmallow=slinearsrvlookup=yes[250]type=friendusername=250secret=dannycallerid="Danny"host=dynamiccontext=demoregister = 100:[EMAIL PROTECTED]/100[sipgate4]type=frienddisallow=allallow=alawallow=ulawfromuser=100authuser=100secret=passwordusername=100host=sipgate.co.ukcontext=demodtmfmode=infofromdomain=sipgate.co.ukinsecure=verynat=yescanreinvite=nocallerid="Danny" lt;0207100My Extensions.conf file contents:[demo]exten = 250,1,Dial(SIP/250,20)exten = 250,2,Voicemail(u250)exten = 250,3,Voicemail(b250)exten = 250,4,Hangupexten = _0207.,1,SetCallerID(""
lt;100|a) ;Outgoingexten = _0207.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,tr)exten = 100,1,Dial(SIP/250,30,tr) ;IncomingAm I have to install any other libraries?Anything wrong in the above configuration?Looking forward to your response. Thanks in advance.Regards,Chandra.
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[asterisk-users] Asterisk+SER help
Hi Friends,I want to setup multiple SIP accounts. How can I do this? I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? 1) Am I need to install SER or OpenSER in my server along with Asterisk?2) If yes, can you please recommond SER or OpenSER?3) I searched in Internet. But, I didn't find good tutorial for this. Can you please tell me a good link for this? Looking forward to your response. Thank you.Regards,Chandra. Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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] Reception Console
Hi,I am interested in test and work with your Reception appliation. Looking forward to your response. Thank you.Regards,Chaandra.Peter Lindquist [EMAIL PROTECTED] wrote: Sure thing, count me inPaul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --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] DID is not working (call is not routing)
Hi Libera,We have an account with Teliax from 7 months. Teliax's service is very good and giving excellent customer support also. But, I observed the below things from Teliax's people.1) Let us assume that you have configured your Teliax account settings with XLite or any other sofphone directly without using Trixbox or Asterisk. After that, if you are facing any problem, they are solving.2) If you configure Teliax account settings with Asterisk or Trixbox, they are facing trouble to solve some technical problems from Trixbox or Asterisk point of view3) Voice quality is very good.Thank you.Regards,Chandra."R.R Libera" [EMAIL PROTECTED] wrote: Hello Chandra, What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in
advance... On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote:Your server seems to be doing exactly what you are telling it
to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created
Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing
'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited
non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] List of VoIP+RJ11 Phones
Hi friends,Thank you to all for response. At last, I got these below links which contains Ethernet port and RJ11 port. http://www.voipsupply.com/product_info.php?products_id=307 http://www.thechewtongroup.com/zultys-zip-4x5.phphttp://gigaset.siemens.com/shc/0,1935,hq_en_0_122378_rArNrNrNrN,00.html Thank you.Regards,Chandra. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --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] VoIP+RJ11 Phone existed?
Hi,I want to buy a phone. That phone must have two ports. One is Ethernet port (to connect to my Asterisk server) and second is RJ11 port (to connect with my traditional PSTN exchange). I searched in internet, but unable to find this phone, which contains both feautre. Can anybody tell me a phone, which consists these both Ethernet and RJ11 ports? Looking forward to your response. Thank you.Regards,Chandra. Get your email and more, right on the new Yahoo.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
[asterisk-users] Anybody using inphonex service?
Hi,I want to register with http://www.inphonex.com VoIP provider. I want to configure my "Trixbox" and "Asterisk" servers with "inphonex". Anybody using this service? Mainly, I want to do three tasks. They are1) Able to make calls to USA2) Able to make international dialing3) Able to receive incoming calls through my DID. (Are they offering DID numbers?)If anybody using this "inphonex" service, please tell me your feedback. Looking forward to your response. Thank you.Regards,Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --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] DID is not working (call is not routing)
Hi William,Thank you for response. Sorry. I forgot to say that am configuring using Trixbox. Can you tell me the solution? Thank you.Regards,Chandra,William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy
Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing
Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing
Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
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Re: [asterisk-users] DID is not working (call is not routing)
Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC.
-- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack --
Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
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[asterisk-users] DID is not working (call is not routing)
Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing
Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing
Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra.
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[asterisk-users] Requirements for Asterisk SER integration
Hi Friends,I would like have Asterisk and SER implementation. I have lot of experience with Asterisk. Can anybody tell me what I have to install to integrate SER with Asterisk? Looking forwrad to your response. Thank you.Regards,Chandra. Get your email and more, right on the new Yahoo.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] New tutorial - peering two * servers using IAX
Hi,Excellent stuff. Thank you.Regards,Chandra.Douglas Garstang [EMAIL PROTECTED] wrote: Alex, Those examples elaborate on the examples supplied with Asterisk, and that's about it. I tried to build a tiered DUNDI model with upstream DUNDi servers that served requests to downstream DUNDi servers that acted as registration servers and used the 'precache' option to send the numbers upstream. I haven't been able to find any docs on this at all. I even posted to the DUNDi list and got bupkiss help. Doug.-Original Message-From: Alex Robar[mailto:[EMAIL PROTECTED]Sent: Wednesday, October 04, 2006 1:13PMTo: Asterisk Users Mailing List - Non-CommercialDiscussionSubject: Re: [asterisk-users] New tutorial - peering two* servers using IAXThere's been a couple of those postedon this list already:http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdfSurethey're for AAH/Trixbox, but the dialplan will work fine with vanilla Asteriskinstalls.Alex On 10/4/06, DouglasGarstang [EMAIL PROTECTED]wrote: How about preparing a step by step guide to DUNDi? Good luck with that though because base DUNDi docs are rarer than periodic element #114 in the known universe.Doug. -Original Message- From: lenz [mailto: [EMAIL PROTECTED]] Sent: Wednesday, October 04, 2006 11:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] New tutorial - peering two * servers using IAX Hi list, today I have been teaching a class on * and have found that many students find it quite hard to understand how setting up IAX peering between two servers may work. So I prepared a little step by step tutorial hoping it might be useful to someone in the future. See it at http://astrecipes.net/index.php?n=204 Comments and corrections are welcome. The site is a wiki, so feel free to modify and improve. l. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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 callerid from PSTN using TDM2400P
Hi,This is Chandra from India. You are from which country? I am asking this because the basic Asterisk setup doesn't recognize callerid in India. I tried to solve this in many ways. But, no use. I think we have to do some modifications in source code. Thank you.Regards,Chandra.Naija Man [EMAIL PROTECTED] wrote: Hello all,Asterisk 1.2.8zaptel 1.2.6Hardware: digium TDM2422PI have a fully configured asterisk system with POTS line for PSTN access. I am not receiving the callerid for incoming calls from the PSTN. I get the following error message. -- Starting simple switch on 'Zap/3-1'Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)...Oct 3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie made mylen
0 (-22) Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: SuccessOct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/3-1My configuration is as below: ***zapata.conf:[channels];usecallerid=yesrestrictcid=nocallerid=asreceivedcidsignalling=bellcidstart=ringhidecallerid=nousecallingpres=yes sendcalleridafter=2ringtimeout=8000echocancel=yesechocancelwhenbridged=yescallprogress=yesbusydetect=yesmusiconhold=defaultuseincomingcalleridonzaptransfer=yesgroup=1context=from-pstn signalling=fxs_kschannel = 1-3extensions.conf:[from-pstn];; Inbound calls from PSTN lineexten = s,1,NoOp(TIMESTAMP: ${TIMESTAMP})exten = s,2,NoOp(CONTEXT: ${CONTEXT}) exten =
s,3,NoOp(CALLERIDNUM: ${CALLERID(number)})exten = s,4,NoOp(CALLERIDNAME: ${CALLERID(name)})exten = s,n,Goto(main-ivr,start,1)*** The variables $CALLERID(number) and $CALLERID(name) always show up empty when a call is received.Any suggestion will be appreciated.Thanks. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Strange problem(Munin-node-1.2.4-7)
Hi,Sorry to post this in this forum.I am new to Trixbox. When I am trying to install Trixbox, I am facing this problem. First I have installed Trixbox ISO image file from a CD. When its rebooting and Asterisk is installing, it is got stucked near this below point:Munin-1.2.4-7Preparing package for installation...0:group munin already present0:user munin already presentMunin-node-1.2.4-7and stopped at this moment. Why this is happening? I tried to installed Trixbox 3 times. But, I faced this problem evertime. Please tell me the problem. Looking forward to your response. Thank you.Regards,Chandra. How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] A Strange doubt and problem
Hi Friends,I got a strange doubt and problem. Is there any problem for SIP protocol, if we configure Intercom (SIP), VoIP (SIP) and PSTN in a single server (that may be Asterisk or Trixbox).My First Experience:Initially I configured Asterisk in a system and created SIP extensions and VoIP with Teliax. It worked very fine for a few days. After that, I configured PSTN (Digium 04B and 20B) and making outgoing and receive incoming calls. After that, SIP protocol was down and unable to make calls to USA using Teliax using SIP. So, I configured IAX2 Teliax account and its working fine now. Why SIP protocol was down?My Second Experience:I have installed Trixbox ISO image in a system and configured as mentioned above. Now, I have faced the same problem. After configuring PSTN only, SIP protocol was dead. What may be the reason?Error Message: When I am making call to USA using Teliax service, it is telling that "All circutis are busy. Please try call later. Thank you".Please share your feelings and experiences. Looking forward to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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: chan_zap.so stopped working after upgrading CentOS
Hi Tony,Dont worry. After upgrading Trixbox, Zaptel won't work. For this, again you need to install Zaptel 1.2.5 files. cd /usr/srcwget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.5.tar.gztar -zxvf zaptel-1.2.5.tar.gzmv zaptel-1.2.5 zaptelcd /usr/src/zaptelmake cleanmake installrebuild_zaptelmodprobe wcfxogenzaptelconfrebootThis will work. Regards,Chandra.Tony Mountifield [EMAIL PROTECTED] wrote: In article <[EMAIL PROTECTED]>,Moises Silva <[EMAIL PROTECTED]> wrote: Why oh why do so many people do all this modprobe stuff manually or in rc.local etc.? If you are running a RedHat / Fedora / CentOS distribution, just do "make config" in the zaptel directory, and it will create a proper startup script in init.d and set up the rc.d links for invocation at boot time. This proper script takes care of loading the modules AND waiting for udev to create the device nodes. Because they are using slackware??? :)But the TWO preceding messages in this branch of the thread both saidthey were running CentOS, and both were doing kludges with individualmodprobes (in one case TWICE!) instead of just using the suppliedscript.CheersTony-- Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] HFC isdn card and bristuff 0.2.0 rc8n
Hi,I am using Trixbox on CentOS. I bought "BT speedway ISDN PCI Card". But, I dont know how to configure this card with Trixbox. I searched a lot in Internet and forums. But, I didn't get any tutorial or any response. You are using this card. So that I am asking to you. Can you please tell me how to configure and install my ISDN card? Looking forward to your response. Thank you.Regards,Chandra.Giordano Grandis [EMAIL PROTECTED] wrote: Hi guys, i have asterisk 1.0.9 with bristuff 0.2.0 rc8n running on a VIA motherboard with processor C3 and i have this kind of problem: during the office time the system work perfectly, but on the next moring, if i try to make an outgoing call i get this message== Primary D-Channel on span 1 down == Primary D-Channel on span 1 upSep 13 08:41:11 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:41:16 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:41:19 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:41:22 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1received TEI check request for TEI = 103received TEI check request for TEI = 103 == Primary D-Channel on span 1 downSep 13 08:41:41 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got S-frame while link down == Primary D-Channel on span 1 down == Primary D-Channel on span 1 down == Primary D-Channel on span 1 down == Primary D-Channel on span 1 downreceived TEI check request for TEI = 103received TEI check request for TEI = 103 == Primary D-Channel on span 1 upSep 13 08:42:00 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:42:02 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:42:03 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1centralino*CLISep 13 08:42:04 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1Sep 13 08:42:05 WARNING[4382]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1received TEI check request for TEI = 103 On locals calls i do not have problem. For * there is not avilable Zap channels. This is my zapata.conf : [channels] language = it switchtype = euroisdnsignalling = bri_cpe_ptmppridialplan = unknownprilocaldialplan = unknownechocancel = yesechocancelwhenbridged = yesechotraining = 10immediate = nogroup = 1callgroup = 1pickgroup = 1musiconhold = defaultcontext = incomingchannel = 1-2 How could y debug this strange situation? Anyone could help me ? Thanks in advance Girodano___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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: Calling Card and Billing
You can try for Trixbox"[EMAIL PROTECTED]" [EMAIL PROTECTED] wrote: Hi all,Let me add to my query.I would prefer to have an Asterisk GUI that has billing calling card solution all in one.Got any suggestions.?Thanks On 12/09/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users,Im looking for recommendations on softwares for calling card implementation and post paid billing services. Please give some recomendatons based on your experiences. A detail of the pros or cons (if any) you faced with it would be highly welcome. thanks in advance.Dan ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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] How to configure Fritz ISDN2 card with Trixbox?
Hi Friends,I have Fritz ISDN2 card and want to configure with Trixbox and Asterisk. I tried to findout for tutorials and installation procedures to install this card. But, I am unable to find. Can anybody please give me a good link or tutorial to install this? Looking forwrad to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] Accounts registered, but call is not going
Hi Friends,1) I am new to Trixbox. 2) First I explain my network architecture. I have a public IP and got internet connection from a ISP. I have connected the internet cable which is coming from ISP to a router. Now, I have connected to Trixbox server to the router. 3) I have assigned static IP (192.168.2.x) to trixbox server and mapped my public IP to trixbox server to access from outside. 4) I have done SIP and IAX port forwarding also with the router.5) Now, I configured my Trixbox server and created two SIP accounts called 101 and 102. 6) Now, I and my friend logged in with XLite softphone with 101 and 102 respectively. 7) If I executed "sip show peers" command, its showing that 101 and 102. 8) Now, If I make a call form 101 from 101, call is not connecting. XLite is telling that "The person you are calling is unavailable, please try again" continuously and XLite screen is displaying that "Call failed:service unavailable"What is the problem? Please tell me a solution. Looking forward to your response. Thank you.Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --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] Incoming call problem-calling part is busy(IPKall)
Hi,I have registered with IPKall ang got the number i.e., 206XXX. When I call to this number, It is telling that "The party you are calling is currently busy". Here I am giving my config details.When I registered with IPKall, I entered these below values:SIP Phone number: 7312567SIP Proxy: voip-co1.teliax.comContents in sip.conf file:[7312567]type=peerdtmfmode=rfc2833context=inboundinsecure=veryhost=voiper.ipkall.comContents in extensions.conf file:[inbound]exten = 7312567,1,Dial(SIP/250,20)include = internalHere, 250 is the SIP account.I have given my total configuration. Please tell me the solution. Looking forward to your response. Thank you.Regards,Chandra. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --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] using SIP to connect remote other VoIP server(Attn:Elpidio)
Hi Elpidio,I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port is not opening. I checked with the below command:# nmap -p5060 192.168.91.22---This is my IP addressand it is showing that port 5060 is closed. How can I enable and open this 5060 port? Really, I am breaking my head with this problem. SIP is not working because of this problem. Please tell me a solution. Looking forward to your reply. Thank you.Regards,Chandra.Elpidio Ramos [EMAIL PROTECTED] wrote: Hi,This is a sample file I am currently using on my server. My server has a public IP address and an internal IP address (duan NIC). It runs Fedora Core 3 running iptables firewall already configured with ports 4569, 5060, 1-2 open (udp and tcp)[general]context=defaultallowguest=norealm=your.hostname.extbindaddr=0.0.0.0bindport=5060externip=your.server.ip.addresssrvlookup=nomaxexpirey=3600disallow=allallow=ulawallow=ilbcallow=gsmmusicclass=defaultlanguage=esrtptimeout=120rtpholdtimeout=300useragent=asterisklocalnet=10.10.10.0/255.255.255.0rtcachefriends=noqualify=yes[311]type=friendregexten=311username=311secret=311callerid="User on extension 311" 311host=dynamicnat=yescanreinvite=no [312] type=friendregexten=312username=312secret=312callerid="User on extension 312" 312host=dynamicnat=yescanreinvite=no tengulre [EMAIL PROTECTED] wrote: How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account? anybody can give me some sample configuration files? thanks a lot! ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Probelm with incoming calls to my DID-Please help me
Hi friends, Thank you to all for your response and cooperation to me. I have a doubt.We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console. Contents in IAX.CONF file:disallow=all allow = ulaw [general] register = teliaxusername:[EMAIL PROTECTED] [teliax] context=telincoming type=friend host=voip-co1.teliax.com auth=md5 secret=teliaxpassword disallow=all allow=ulaw allow=alaw allow=gsm
Contents in Sip.conf file: [105] type=friend username=105 secret=ravi callerid="RaviKanth" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED] [107] type=friend username=107 secret=suresh callerid="Suresh" host=dynamic context=administration canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED] Contents in Extensions.conf file:[telincoming] exten = 303xxx, 1, Answer() exten = 303xxx, n, Wait,2 exten = 303xxx, n, Goto(incoming,s,1) include = internal include = incoming [incoming]
exten = s,1,Wait(3) exten = s,n,Answer exten = s,n,SetMusicOnHold(default) exten = s,n,Set(TIMEOUT(digit)=5) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Background(/tmp/virg2) exten = s,n,Goto(s,1) exten = s,n,Hangup() include = internal [internal] exten = 105,1,SetMusicOnHold(default) exten = 105,2,Dial(SIP/105,7,t,m,T) exten = 1605,1,VoiceMailMain([EMAIL PROTECTED]) exten = 105,3,VoiceMail([EMAIL PROTECTED]) exten = 105,4,Hangupexten = 107,1,SetMusicOnHold(default) exten = 107,2,Dial(SIP/107,7,t,m,T) exten = 1607,1,VoiceMailMain([EMAIL PROTECTED]) exten = 107,3,VoiceMail([EMAIL PROTECTED]) exten = 107,4,Hangup[uscall] exten = _1XX,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr) [manager] include = uscall include = internalThe error message on Asterisk console: *CLI -- Executing Dial("SIP/105-007951e0", "IAX2/[EMAIL PROTECTED]/1303xxx|30|tr") in new stack-- Called [EMAIL PROTECTED]/1303xxx-- Call accepted by 207.174.202.2 (format ulaw)-- Format for call is ulaw-- IAX2/teliax-1 is ringing-- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0-- IAX2/teliax-1 is ringing-- IAX2/teliax-1 is busy-- Hungup 'IAX2/teliax-1'== Everyone is busy/congested at this time (1:1/0/0)== Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'What is the
problem? Can you please tell me the solution. Looking forward to your response. Thank you. Regards, Chandra.
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
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Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Hi,I am Chandra. I have a doubt related to ports. I have seen my port 5060 status with nmap command and it is showing that 5060 blocked. Afterthat, I stopped firewall also. After stopping the firewall also, it is showing the 5060 port is blocked. Can I need to restart the linux system from boot to take the effect? Please tell me how to open 5060 port? Looking forward to your response. Thank you.Regards,Chandra.Steven Ringwald [EMAIL PROTECTED] wrote: Elpidio Ramos wrote: Bob, I get the same answer you get when using netstat -an When I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references) target prot opt source destination ACCEPT all -- anywhere anywhereACCEPT icmp -- anywhere anywhereicmp any ACCEPT ipv6-crypt-- anywhere anywhereACCEPT ipv6-auth-- anywhere anywhereACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353 ACCEPT udp -- anywhere anywhereudp dpt:ipp ACCEPT all -- anywhere anywherestate RELATED,ESTABLISHED ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:ssh ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:http REJECT all -- anywhere anywhere reject-with icmp-host-prohibited I assume this indicates port 5060 is restricted?Yep.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --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] Probelm with incoming calls to my DID-Please help me
Hi Marco,Thank you for your response. I tried as suggested by you. But, Its not working. I also posted this question to Teliax people. They told me as below:You are not fully registered to us. Your IP is not shown on a show peercommand. There is something missing in your set up. You mustregister the IP to complete the call traffic.Now, what I have to do? Can I configure any otherfiles in Asterisk? Please do needful. Looking forward to your response. Thank you.Regards,Chandra.Marco Mouta [EMAIL PROTECTED] wrote: Hi,Please read bellow:On 9/1/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi friends, Thank you to all for your response and cooperation to me. I have a doubt.We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console. Contents in IAX.CONF file:disallow=all allow = ulaw [general] register = teliaxusername:[EMAIL PROTECTED]
[teliax] context=telincoming type=friend host= voip-co1.teliax.com auth=md5 secret=teliaxpassword disallow=all allow=ulaw allow=alaw allow=gsmContents in Sip.conf file: [105] type=friend username=105 secret=ravi callerid="RaviKanth" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED] [107] type=friend username=107 secret=suresh callerid="Suresh" host=dynamic context=administration canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED] Contents in Extensions.conf file:[telincoming] exten = 303xxx, 1, Answer() exten = 303xxx, n, Wait,2 exten = 303xxx, n, Goto(incoming,s,1) You need to inser "_" before a pattern so asterisk can try to match it: exten = _303xxx, 1, Answer()exten = _303xxx, n, Wait,2 exten = _303xxx, n, Goto(incoming,s,1) Should solve your problem!Also only as debug you can try _X. Pls tell me if it solved your problem. include = internal include = incoming [incoming] exten = s,1,Wait(3) exten =
s,n,Answer exten = s,n,SetMusicOnHold(default) exten = s,n,Set(TIMEOUT(digit)=5) exten = s,n,Set(TIMEOUT(response)=10) exten = s,n,Background(/tmp/virg2) exten = s,n,Goto(s,1) exten = s,n,Hangup() include = internal [internal] exten = 105,1,SetMusicOnHold(default) exten = 105,2,Dial(SIP/105,7,t,m,T) exten = 1605,1,VoiceMailMain( [EMAIL PROTECTED]) exten = 105,3,VoiceMail([EMAIL PROTECTED]) exten = 105,4,Hangupexten = 107,1,SetMusicOnHold(default) exten = 107,2,Dial(SIP/107,7,t,m,T) exten = 1607,1,VoiceMailMain( [EMAIL PROTECTED]) exten = 107,3,VoiceMail([EMAIL PROTECTED]) exten = 107,4,Hangup[uscall] exten = _1XX,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr) [manager] include = uscallinclude = internalThe error message on
Asterisk console: *CLI -- Executing Dial("SIP/105-007951e0", "IAX2/[EMAIL PROTECTED] /1303xxx|30|tr") in new stack-- Called [EMAIL PROTECTED]/1303xxx-- Call accepted by 207.174.202.2 (format ulaw)-- Format for call is ulaw-- IAX2/teliax-1 is ringing-- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0-- IAX2/teliax-1 is ringing -- IAX2/teliax-1 is busy-- Hungup 'IAX2/teliax-1'== Everyone is busy/congested at this time (1:1/0/0)== Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'What is the problem? Can you please tell me the solution.
Looking forward to your response. Thank you. Regards, Chandra. Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or
update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Get on board. You're invited to try the new Yahoo! Mail. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra."Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] wrote: Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month?Most providers have unlimited minutes on the plans that are not flat rate. i.e. you can use as many mins as you want at 2/cents/min.If you mean "unlimited for a flat monthly fee" there is nobody out there stupid enough to offer that service, or, if they are, they don't stay in business.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] does anyone offer truly unlimited voip in the US
Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra."Steven M. Sawczyn" [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define "unlimited" as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --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] Connecting two asterisk servers
Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] Nobody is responding. Why? (Implement music on transfer)
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] How can I implement Music on Call Transfer?
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. Get your email and more, right on the new Yahoo.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
[asterisk-users] How can I implement Music on Call Transfer?
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --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] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Dear Leo,As you said, I have tried using dtmf and in different values. But, no reuslt. Finally, I knew that Basic Asterisk setup doesn't recognize callerid in India. To get callerid in India, we have to do some modifications in chan_zap.c source file. But, I dont know what modifications I have to do? Do you have any Idea about these modifications in source code? Can you please tell me. Looking forward to your response. Thank you.Regards,Chandra.Leo Ann Boon [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Leo, Thank you for your quick response. In Internet, I came to know that 1) In India, we have to give dtmf and ring for cidsignallling and cidstart respectively.Have you tried settingcidsignalling=dtmf 2) Default Asterisk setup doesn't
recognise callerid in India. To recognize callerid in India, we have to do or change some modifications in chan_zap.c source file. Is it right?If cidsignalling=dtmf won't work then you might have to consider invasive surgery on chan_zap. :) 3) Please open the below link and see the values for India. http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf The zaptel driver has tone definition for india. In /etc/zaptel.conf:loadzone=indefaultzone=in Here I am giving the error messages on Asterisk console. *CLI -- Starting simple switch on 'Zap/1-1' Aug 18 14:53:13 ERROR[15499]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-16) Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6131 ss_thread: CallerID returned with
error on channel 'Zap/1-1' -- Executing Wait("Zap/1-1", "10") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing NoOp("Zap/1-1", " 18082006-14:53:24") in new stack -- Executing NoOp("Zap/1-1", "CallerID is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Name is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Number is ") in new stack -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("Zap/1-1", "/tmp/virg2") in new stack -- Playing '/tmp/virg2' (language 'en') == CDR updated on Zap/1-1 -- Executing Dial("Zap/1-1", "SIP/105|15|t|12") in new
stack -- Called 105 -- SIP/105-00798410 is ringing -- Nobody picked up in 15000 ms -- Executing VoiceMail("Zap/1-1", "u105") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (incoming, 105, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Please tell me. Looking forward to your response. Thank you. Regards, Chandra. *//*___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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RE: [asterisk-users] CallerID is not displaying for my incoming calls
Hi Rushowr,Thank you for your response. As you said, I executed these below lines:exten = s,n,Verbose(2|CallerID info received: ${CALLERID(all)}) ; shows CID info exten = s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID presentationAnd Asterisk is showing this below error on console:Executing Verbose("Zap/1-1", "3|CallerID info received: "" ") in new stackCallerID info received : "" Executing Verbose("Zap/1-1", "3|Presentation setting: 0") in new stackPresentation setting: 0As per my knowledge, I have to do some modifications in chan_zap.c file to get callerid in India. But, I dont know what modifications i have to do? Can you pleaes tell me.Looking forward to your reply. Than you.Regards,Chandra.Rushowr [EMAIL PROTECTED] wrote: Chandra, Unfortunately, I can't help you too much, because I've not worked a lot with Zap. However, this message: Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Seems
interesting...My guess is that the callerid information is corrupted or something, because it's a negative value, not a 0 or positive. Possibly you have your CID Signalling set to the wrong value... One thing you could try just to get a better idea of what (if anything) is actually read from the callerid and what the presentation is set to, is to modify the your dialplan to output the data to your console (I use verbose 2 so I don't have to read the extra info: [incoming]exten = s,1,Wait(4)exten = s,n,Answer exten = s,n,Verbose(2|CallerID info received: ${CALLERID(all)})
; shows CID info exten = s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID presentationexten = s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader Hope this is helpful in some way... Rushowr From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of CrazyBoySent: Friday, August 18, 2006 1:14 AMTo: AsteriskUsers Mailing List - Non-Commercial DiscussionSubject: RE:[asterisk-users] CallerID is not displaying for my incomingcalls Hi Rushowr,Thank you for response.Here I am givingmy config files and error message. Please see it.zaptel.conf contents:loadzone =usdefaultzone=usfxsks=1-4zapata.conf
contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel= 1sip.confcontents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.confcontents:[incoming]exten = s,1,Wait(4)exten =s,n,Answerexten = s,n,SetMusicOnHold(default)exten
=s,n,Set(TIMEOUT(digit)=5)exten =s,n,Set(TIMEOUT(response)=10)exten =s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten =s,n,Hangup()include = leader[leader]exten =105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten =105,3,Voicemail(b105)exten = 105,4,Hangupexten =_9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten= _5,1,Dial(Zap/1/${EXTEN:1}) ;Local Landlineinclude = internal[internal]exten = 105,1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I amgetting this below error message on Asterisk console:Error Message:Aug 17 19:45:41ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0(-8)Aug 17 19:45:41 WARNING[10449]:
chan_zap.c:6087 ss_thread:CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]:chan_zap.c:6131 ss_thread: CallerID returned with error on channel'Zap/1-1'Please tell me the solution. Looking forward to your kindresponse. Thank you.Regards,Chandra.Rushowr[EMAIL PROTECTED] wrote: What's the Dial command being used to pass the call to the Softphones? From:
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of CrazyBoySent: Wednesday, August 16, 2006 3:23 AMTo:[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussionSubject: Re: [asterisk-users] CallerID is notdisplaying for my incoming calls Hi,As you said, I have changed my configurations. But,callerid is not displaying. What I have to do? Please tellme.ThanksRegards,Chandra.Rich Adamson[EMAIL PROTECTED] wrote:Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line
directly to first port. I am making outgoin
Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Hi Leo,Thank you for your quick response. In Internet, I came to know that1) In India, we have to give dtmf and ring for cidsignallling and cidstart respectively. 2) Default Asterisk setup doesn't recognise callerid in India. To recognize callerid in India, we have to do or change some modifications in chan_zap.c source file. Is it right? 3) Please open the below link and see the values for India. http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf Here I am giving the error messages on Asterisk console.*CLI -- Starting simple switch on 'Zap/1-1'Aug 18 14:53:13 ERROR[15499]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-16)Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6087 ss_thread:
CallerID feed failed: SuccessAug 18 14:53:13 WARNING[15499]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Wait("Zap/1-1", "10") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing NoOp("Zap/1-1", " 18082006-14:53:24") in new stack -- Executing NoOp("Zap/1-1", "CallerID is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Name is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Number is ") in new stack -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("Zap/1-1", "/tmp/virg2")
in new stack -- Playing '/tmp/virg2' (language 'en') == CDR updated on Zap/1-1 -- Executing Dial("Zap/1-1", "SIP/105|15|t|12") in new stack -- Called 105 -- SIP/105-00798410 is ringing -- Nobody picked up in 15000 ms -- Executing VoiceMail("Zap/1-1", "u105") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (incoming, 105, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1'Please tell me. Looking forward to your response. Thank you.Regards,Chandra.Leo Ann Boon
[EMAIL PROTECTED] wrote: Please see my response in-line.Crazy Boy wrote: Hi Leo, Thank you for your response. I am answering for your questions. Q) As El mentioned - did you actually subscribe for callerid? Most telcos will charge it as a VAS(Value Added Service). Ans) Yes. You are right. I have already subscribed for callerid and tested with an analog phone with callerid instrument. Q) Check the format of the Caller ID provided by your telco - bell,v23 or dtmf? Ans) I dont know how to check my caller id format provided by our provider. Can you please explain how to check my caller id format?You have to ask your provider or check with your local regulator. From your error log, I'm fairly certain Asterisk is not detecting the
caller-id. So, it's either its not sent by the telco or it's in the wrong format. US uses the bell format, UK and many Commonwealth countries use v23 while some European countries use DTMF. I'm not familar with India, but I think it's not bell (Asterisk's default). Q) Check when is Caller ID sent, in some places it's between 1st and 2nd. Other between 2nd and 3rd. You need to Wait(?) as El suggested to wait for it to be sent. Ans) As you said, I put the Wait(4) statement in extensions.conf file in [incoming] context. But, callerid is not displaying. Q) Is there any reason you're using US tones instead of India? Ans) No reason. Is there any effect on getting callerid, if i use like this.It's not important for caller id, but may create other issues like hangup. Q) Is your line really a kewlstart line? I think it should more likely be
loopstart. Ans) Frankly, I dont know what is kewlstart? Can you please tell me.Please see this http://ourproject.org/docman/view.php/116/144/faq.html#TDM%20%20Analog_1kewlstart is pretty much exclusive to Asterisk (and some channel banks) to provide disconnect supervision. For telco analog lines, it's usually loopstart or groundstart. kewlstart is based on loopstart so you should be able to place and receive calls, but you'll run into other issues.Word of advice: Please get hold of a copy of your local telecommunication signaling standards. Without that, it's like navigating a ship in the dark without a map.Regards.Leo___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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[asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Hi,Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as "Asterisk" in my softphones (XLite).Here I am giving my config files and error message. Please see it.zaptel.conf contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel = 1sip.conf contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf contents:[incoming]exten = s,1,Wait(4)exten = s,n,Answerexten =
s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude = internal[internal]exten = 105, 1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8)Aug 17 19:45:41 WARNING[10449]:
chan_zap.c:6087 ss_thread: CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your kind response. Thank you.Regards,Chandra.
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Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Hi Leo,Thank you for your response. I am answering for your questions.Q) As El mentioned - did you actually subscribe for callerid? Most telcos will charge it as a VAS(Value Added Service).Ans) Yes. You are right. I have already subscribed for callerid and tested with an analog phone with callerid instrument.Q) Check the format of the Caller ID provided by your telco - bell,v23 or dtmf?Ans) I dont know how to check my caller id format provided by our provider. Can you please explain how to check my caller id format?Q) Check when is Caller ID sent, in some places it's between 1st and 2nd. Other between 2nd and 3rd. You need to Wait(?) as El suggested to wait for it to be sent.Ans) As you said, I put the Wait(4) statement in extensions.conf file in [incoming] context. But, callerid is not displaying.Q) Is there any reason you're using US tones instead of India?Ans) No reason. Is there any effect on getting
callerid, if i use like this.Q) Is your line really a kewlstart line? I think it should more likely be loopstart.Ans) Frankly, I dont know what is kewlstart? Can you please tell me.Here I am giving my config files and error message. Please see it.zaptel.conf contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel = 1sip.conf contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf contents:[incoming]exten = s,1,Wait(4)exten = s,n,Answerexten = s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude =
internal[internal]exten = 105, 1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8)Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your kind response. Thank you.Regards,Chandra.Leo Ann Boon [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi, Here I am posting my problem. I am getting this problem since 8
days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as "Asterisk" in my softphones (XLite).A few things:a. As El mentioned - did you actually subscribe for callerid? Most telcos will charge it as a VAS(Value Added Service).b. Check the format of the Caller ID provided by your telco - bell,v23 or dtmf?c. Check when is Caller ID sent, in some places it's between 1st and 2nd. Other between 2nd
and 3rd. You need to Wait(?) as El suggested to wait for it to be sent.Other things I see in your config:a. Is there any reason you're using US tones instead of India?b. Is your line really a kewlstart line? I think it should more likely be loopstart.Leo.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [asterisk-users] CallerID is not displaying for my incoming calls
Hi Rushowr,Thank you for response.Here I am giving my config files and error message. Please see it.zaptel.conf contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel = 1sip.conf
contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf contents:[incoming]exten = s,1,Wait(4)exten = s,n,Answerexten = s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude = internal[internal]exten = 105, 1,
Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8)Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your kind response. Thank you.Regards,Chandra.Rushowr [EMAIL PROTECTED] wrote: What's the Dial command being used to pass the call to the Softphones? From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of CrazyBoySent: Wednesday, August 16, 2006 3:23 AMTo:[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussionSubject: Re: [asterisk-users] CallerID is not displayingfor my incoming calls Hi,As you said, I have changed my configurations. But,callerid is not displaying. What I have to do? Please tellme.ThanksRegards,Chandra.Rich Adamson
[EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files. zaptel.conf file contents: loadzone = us defaultzone=us fxsks=1-4 zapata.conf file contents: [channels] context=incoming signalling=fxs_ks
busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes hidecallerid=no echocancel=yes transfer=yes immediate=no group=1 callgroup=9 pickupgroup=9 channel = 1The above entries appear to be reasonable and correct. If you have not properly set rxgain and txgain, it "could" impact callerid. If those gains are too high/low, asterisk will not properly read the callerid data when sent to you. extensions.conf file contents: [incoming] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten =
s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) include = leader Got event 18 (Ring Begin)... Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead.The above two WARNING statements are telling you that either you are copying those exten= statements from someone's old config files, or, you haven't read the asterisk documentation. The message is telling you that your statement "exten = s,3,DigitTimeout,5" should be replaced with the Set(TIMEOUT(digit)=timeout) syntax. Your statements are still
executing properly today, but the next time you upgrade asterisk code, they are likely to fail due to the old syntax not being supported.Try 'show function TIMEOUT' from your CLI and read it. What I have to do to display the PSTN caller number on my softphones? Please tell me. Looking
Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Hi Flynn,Thank you for response. As you asked, I got subscribed for getting callerid and tested with callerid phone also. Here I am giving my config files and error message. Please see it.zaptel.conf contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel = 1sip.conf
contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf contents:[incoming]exten = s,1,Wait(4)exten = s,n,Answerexten = s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude = internal[internal]exten = 105, 1,
Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8)Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your kind response. Thank you.Regards,Chandra.El Flynn [EMAIL PROTECTED] wrote: Crazy Boy wrote:Hi, Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and
looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as "Asterisk" in my softphones (XLite). When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console: Error Message: Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID
returned with error on channel 'Zap/1-1' Please tell me the solution. Looking forward to your kind response. Do you actually _HAVE_ caller ID on that PSTN line?Flynn___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] CallerID is not displaying for my incoming calls
Hi,As you said, I have tested. But, still callerid is not displaying. On the console, Asterisk is giving below error: *CLI Aug 14 15:09:58 ERROR[27056]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 14 15:09:58 WARNING[27056]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 14 15:09:58 WARNING[27056]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your response. Thanks Regards,Chandra.Ira [EMAIL PROTECTED] wrote: At 02:14 AM 8/14/2006, you wrote:We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files.[incoming]exten = s,1,wait(2)exten = s,n,Answerexten = s,n,SetMusicOnHold(default)exten = s,n,DigitTimeout,5exten = s,n,ResponseTimeout,10exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)include = leaderWhat I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.When I had this problem, adding a wait() in front of the answer cured the problem. I have the same TDM04 card and we get callerid no problem now.Ira ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --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] CallerID is not displaying for my incoming calls
Hi,As you said, I have changed my configurations. But, callerid is not displaying. What I have to do? Please tell me.ThanksRegards,Chandra.Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files. zaptel.conf file contents: loadzone = us defaultzone=us fxsks=1-4 zapata.conf file contents: [channels] context=incoming signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes hidecallerid=no echocancel=yes transfer=yes immediate=no group=1 callgroup=9 pickupgroup=9 channel = 1The above entries appear to be reasonable and correct. If you have not properly set rxgain and txgain, it "could" impact callerid. If those gains are too high/low, asterisk will not properly read the callerid data when sent to you. extensions.conf file contents: [incoming] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) include = leader Got event 18 (Ring Begin)... Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead.The above two WARNING statements are telling you that either you are copying those exten= statements from someone's old config files, or, you haven't read the asterisk documentation. The message is telling you that your statement "exten = s,3,DigitTimeout,5" should be replaced with the Set(TIMEOUT(digit)=timeout) syntax. Your statements are still executing properly today, but the next time you upgrade asterisk code, they are likely to fail due to the old syntax not being supported.Try 'show function TIMEOUT' from your CLI and read it. What I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Problems with incoming authentication
Hi,I am Chandra from India. We have Installed Asterisk in our organization. We want to buy a VoIP plan to make calls to US. I have some doubts. Please clarify. 1) How is the Voicepulse service?2) Is Voicepulse working fine with Asterisk?3) Can I configure Voicepulse easily with Asterisk?4) From Teliax and Voicepulse, Which is offering better service?Looking forward to your response. Thank you.Regards,Chandra.David Freeman [EMAIL PROTECTED] wrote: Sometimes I can receive a call to my DID, but sometimes it just rings and rings and I see these messages in the full log:[Aug 14 23:32:12] DEBUG[3556] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:12] NOTICE[3556] chan_iax2.c: Host 64.61.93.87 failed to authenticate as voicepulse[Aug 14 23:32:13] DEBUG[3550] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:13] NOTICE[3550] chan_iax2.c: Host 64.61.93.90 failed to authenticate as voicepulseUsually, when this happens, I can immediately re-dial the DID and * receives the call and sends it on the dialplan.I've tried configuring my IAX2 user details to no use rsa and no key, but the problem persists. I've tried to delete my IAX2 trunks to just use the SIP ones and I get the same problem...in fact, if I only have SIP trunks, I can't receive any calls to the DID.I'm using Asterisk SVN-trunk-r39753M currently, updated today. Any help would be appreciated, I'm running out of ideas. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --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] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi,Thank you very much for your patience to give solutions for me. Today is holiday for us because of our Independance day. Tomorrow I will do and check as suggested by you and let you know. Once again, Thank you.Regards,Chandra.Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: I struggled with one provider for a long time until finally realizing my username on their site was not my username that I was supposed to be using in sip configuration. Make sure you are using the right username and password. However, it would seem that you would not be able to make an outgoing call using the wrong username/password combination. One thing I have not seen in your posts is your firewall information. Your firewall may be setup to allow outgoing connections, but not incoming. I would not depend on info from a provider. You may very well be registering with them, but your firewall may be blocking the incoming call. If you think you have no firewall, check again. IPTABLES might have loaded itself and it may be blocking. Try: service iptables stop and then try the incoming call again. I've been burned twice due to this. Something has changed in the way I configure my linux boxes, and for some reason iptables is starting. On 8/14/06, Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi, Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is "My Asterisk server doesn't register with Teliax". Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth =xyz.abc:[EMAIL PROTECTED] Double check the above two statements to ensure the userid and passwordare exactly those provided to you by teliax. There is nothing else inyour config that impacts the register statement with the exception of nat'ing.It would appear from your other config statements that asterisk might belocated behind a firewall or nat box. If so, read the documentation onthat, and look at the asterisk/configs/sip.conf.sample file. Specifically the section on "NAT SUPPORT".You might also want to read more about using the diagnostic toolsavailable to you within asterisk. Setting verbose and/or debug to a highnumber and copy/paste the CLI output associated with the problem. Or, start using the CLI with something like:asterisk -rvv [teliax-incoming] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) The above has nothing to do with registering with teliax, but you do notwant to "answer" a call before ringing the sip phone. Take thatstatement out of there. When the sip phone answers an incoming call, asterisk will automatically send the answer to teliax.___--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 -- Lacy MooreAspendora, Inc. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --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] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi,Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is "My Asterisk server doesn't register with Teliax". Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me.SIP.CONF contents:[general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] [teliax] context=teliax-incoming type=friend username=xyz.abc user=xyz.abc host=voip-co1.teliax.com secret=xxx
insecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm[105] type=friend username=105 secret=rani callerid="Ranikumar" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=allEXTENSIONS.CONF contents:[leader] exten = 105,1,Dial(SIP/105,15) exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)[teliax-incoming] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)Please tell me the solution. Looking forward to your response.Thank you.Regards, Chandra. Rich Adamson [EMAIL PROTECTED]
wrote: Thank you for your response. As you said, I changed the context "default" to "general". Now, 1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension.No one can guess at the above without you providing something from the CLI to indicate what is going on. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering with Teliax, How is it possible?The register statement is "only" used to inform teliax that your system is on line, can be reached at the IP address determined via the register effort, and if you have something at the end of the register statement (like /1234) teliax will send that "1234" extension in their effort to
complete a call "to" your asterisk system.The register statement has nothing to do with you initiating calls to them. 3) How can I know that whether my Asterisk server is registered with Teliax or not? From the CLI, do a 'sip show registry' and it will tell you. If there is an entry shown, its registered. 4) Registering with Teliax is different for outgoing and incoming or the same?As mentioned above, the registering is only used to inform the teliax boxes "how to reach you".Your outgoing calls to teliax use the definitions you provided in the [teliax] context, just exactly like you copied them from the teliax.com web site. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi, My user name is : rudy.pandya Thank you.Sharon Lim [EMAIL PROTECTED] wrote: I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is " My Asterisk server doesn't register with
Teliax". Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me.SIP.CONF contents:[general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] [teliax] context=teliax-incoming type=friend username=xyz.abc user=xyz.abc host=
voip-co1.teliax.com secret=xxxinsecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm[105] type=friend username=105 secret=rani callerid="Ranikumar" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=allEXTENSIONS.CONF contents:[leader] exten = 105,1,Dial(SIP/105,15) exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)[teliax-incoming]exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)Please tell me the solution. Looking forward to your response.Thank you.Regards, Chandra.
Rich Adamson [EMAIL PROTECTED] wrote: Thank you for your response. As you said, I changed the context "default" to "general". Now, 1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension.No one can guess at the above without you providing something from the CLI to indicate what is going on. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering with Teliax, How is it possible?The register statement is "only" used to inform teliax that your system is on line, can be reached at the IP address determined via the
register effort, and if you have something at the end of the register statement (like /1234) teliax will send that "1234" extension in their effort to complete a call "to" your asterisk system.The register statement has nothing to do with you initiating calls to them. 3) How can I know that whether my Asterisk server is registered with Teliax or not? From the CLI, do a 'sip show registry' and it will tell you. If there is an entry shown, its registered. 4) Registering with Teliax is different for outgoing and incoming or the same?As mentioned above, the registering is only used to inform the teliax boxes "how to reach you".Your outgoing calls to teliax use the definitions you provided in the [teliax] context, just exactly like you copied them from the teliax.com web
site. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
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[asterisk-users] CallerID is not displaying for my incoming calls
Hi Friends,We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files.zaptel.conf file contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf file contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nogroup=1callgroup=9pickupgroup=9channel = 1sip.conf file contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf file contents:[incoming]exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,10exten =
s,5,Background(/tmp/virg2)exten = s,6,Goto(s,1)include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude = internal[internal]exten = 105, 1, Dial(SIP/105,15)My Asterisk console displayed these below messages, when a call comes from PSTN:Got event 18 (Ring Begin)... Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use
Set(TIMEOUT(response)=timeout) instead.What I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi, Thank you for your response. As you said, I changed the context "default" to "general". Now, 1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering with Teliax, How is it possible? 3) How can I know that whether my Asterisk server is registered with Teliax or not? 4) Registering with Teliax is different for outgoing and incoming or the same? Please tell me. Looking forward to your response. Thank you. Regards, Chandra. hugolivude [EMAIL PROTECTED] wrote: Note that you have: [teliax] context=defaultbut you do not have a "default" context in extensions.conf
for this.Change the above to: [teliax] context=general**OR** in extensions.conf change [general] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)to: [default exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)On 8/11/06, Crazy Boy wrote: Hi friends, We have installed Asterisk in our organization. We registered with Teliax and got our DID number. We are making calls to USA successfully through Asterisk. We are making outgoing calls to US. But, we are unable to receive incoming calls to our DID number. When I executed the "sip show peers" command, it is showing that my Asterisk server is registered and displaying Teliax IP address also. I checking by doing ping to voip-co1.teliax.com. Pinging is
also going fine. Here I am giving the configuration files. Please tell me a solution. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] [teliax] context=default type=friend username=xyz.abc user=xyz.abc host=voip-co1.teliax.com secret=xxx insecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm [105] type=friend username=105 secret=rani callerid="Ranikumar" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all EXTENSIONS.CONF contents: [leader] exten = 105,1,Dial(SIP/105,15) exten
= 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) [general] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) Please tell me the solution. Looking forward to your response. Thank you. Regards, Chandra. Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi Ram,I have given the onfiguration files in the last of this mail. Please read that. I registered with teliax and making calls to US using Teliax. As you said, I executed the command "sip show registry". But, Its not showing any registered users. But, how i am doing outgoing calls to US?Looking forward to your response.ThanksRegards,Chandra.ram [EMAIL PROTECTED] wrote: Hi Chandra You check in the console asterisk -r sip show regis will show you the account is Registered with your Voip Provider or not If not try add in the conf file register=account:[EMAIL PROTECTED]/account
Ram On 8/12/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,Thank you for your response. As you said, I changed the context "default" to "general". Now,1) When I am making call to our DID, its ringing. But, call is not transferring to 105 extension. 2) Teliax people told me that my Asterisk server doesn't register with Teliax. But, I am making calls to US using Teliax. Without registering with Teliax, How is it possible?3) How can I know that whether my Asterisk server is registered with Teliax or not? 4) Registering with Teliax is different for outgoing and incoming or the same?Please tell me. Looking forward to your
response.Thank you.Regards,Chandra. hugolivude [EMAIL PROTECTED] wrote: Note that you have: [teliax] context=defaultbut you do not have a "default" context in extensions.conf for this.Change the above to: [teliax] context=general**OR** in extensions.conf change [general] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15)to: [default exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) On 8/11/06, Crazy Boy wrote: Hi friends, We have installed Asterisk in our organization. We registered with Teliax and got our DID number. We are making calls to USA successfully through Asterisk. We are making outgoing calls to US. But, we are unable to receive incoming calls to our DID number. When I executed the "sip show peers" command, it is showing that my Asterisk server is registered and displaying Teliax IP address also. I checking by doing ping to voip-co1.teliax.com. Pinging is also going fine. Here I am giving the configuration files. Please tell me a solution. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] [teliax] context=default type=friend username=xyz.abc user=xyz.abc host= voip-co1.teliax.com secret=xxx insecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm [105] type=friend username=105 secret=rani callerid="Ranikumar" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all
EXTENSIONS.CONF contents: [leader] exten = 105,1,Dial(SIP/105,15) exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) [general] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) Please tell me the solution. Looking forward to your response. Thank you. Regards, Chandra. Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Everyone is raving about the all-new Yahoo! Mail Beta. ___--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 listTo UNSUBSCRIBE or update options visit:
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[asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi friends, We have installed Asterisk in our organization. We registered with Teliax and got our DID number. We are making calls to USA successfully through Asterisk. We are making outgoing calls to US. But, we are unable to receive incoming calls to our DID number. When I executed the "sip show peers" command, it is showing that my Asterisk server is registered and displaying Teliax IP address also. I checking by doing ping to voip-co1.teliax.com. Pinging is also going fine. Here I am giving the configuration files. Please tell me a solution. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] [teliax] context=default type=friend username=xyz.abc user=xyz.abc host=voip-co1.teliax.com secret=xxx
insecure=very canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm [105] type=friend username=105 secret=rani callerid="Ranikumar" host=dynamic context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all EXTENSIONS.CONF contents: [leader] exten = 105,1,Dial(SIP/105,15) exten = 105,2,Voicemail(u105) exten = 105,3,Voicemail(b105) exten = 105,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) [general] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) Please tell me the solution. Looking forward to your response. Thank you. Regards, Chandra.
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[asterisk-users] How to connect Snom softphone from my home?
Hi Friends, We have installed "Asterisk" in our office and using it successfully. I have given public IP to our Asterisk server. We are using Snom360 5.3 softphone for communication. I tried to connect to our Asterisk server with my Snom360 5.3 softphone from my house. But, it is not connecting. How can I connect from my house to my Asterisk server through Snom softphone?This is very urgent. Looking forward to your kind response. Thank you.Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] How to configure NOKIA N70 with Asterisk?
Hi,I also have Nokia N70. After seeing your mail, I also tried to configure my N70 to connect to my Asterisk. But, I am unable to find the "SIP settings" option in my mobile (Tools-Settings-Connection-SIP settings). What I have to do? Looking forward to your response. Thank you.Regards,Chandra.Jean-Yves Avenard [EMAIL PROTECTED] wrote: HiOn 8/1/06, FaberK <[EMAIL PROTECTED]> wrote: Hi folks, I got an N70. Any lynks for the voip/sip configuration? Thanks .:FaberK:.they aren't hard to find !this one works for me:http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.htmlOne note of warning :the Nokia will not work if behing NAT ... I've tried everything butI've never managed to get it to work unless the Nokia had a public IPaddress or was on the same subnet as the asterisk server.Be interested to know if you can find a way around thisJY___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta.___ --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] cannot received calls in pstn line
Hi,I also faced same problem initially. Please write down your configuration.Regards,ChandraLito Lampitoc [EMAIL PROTECTED] wrote: sorry for my english, but here' s the scenario:I have a 1 FXO and 1 FXS. when my telephone (direct line) is connected to the FXO, I cannot receive an incoming call. Since I am in an office with conventional PBX, I tried to connect one local line (local to PBX) to the FXO and made a call from other direct lines (outside the office) and it works! brandon, i'll try your suggestion.thanks. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
Hi Friends, I am Chandra from India. Thank you for your cooperation and for clear my doubts. Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. When I am making calls from outside to my PSTN number, sometimes Asterisk receving that call and sometimes, its not receiving. Why? Here I am giving my configuration of my files. ZAPTEL.CONF contents: loadzone = us defaultzone=us fxsks=1,2,3,4 ZAPATA.CONF contents: [channels] context=tutorial signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes echocancel=yes transfer=yes immediate=no group=1 channel = 1 SIP.CONF contents: [300] type=friend username=300 secret=server callerid="Server" host=dynamic context=tutorial [general] port=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ulaw allow=alaw EXTENSIONS.CONF contents: TRUNK=Zap/g1 TRUNK=Zap/g2 [tutorial] exten = s,1,Dial(SIP/350,30) exten = s,n,Voicemail(350) exten = s,n,Hangup exten = 300,1,Dial(SIP/300,15) exten = 300,2,Voicemail(u300) exten = 300,3,Voicemail(b300) exten = 300,4,Hangup What is the solution? Please tell me. Looking forward to your response. Thank you. Regards, Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] Strange error
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything. My intercom is also working fine always. What is this error? Please tell me the solution. Looking forward to your response. ThanksRegards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --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] SIP is not working sometimes. IAX is working fine. Why?
Hi, We are using "Asterisk" in our office and using "XLite" as softphone and your service for making calls to USA. When I am using SIP, Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, it is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything or any modifications. My intercom is also working fine always. What is this error? Please tell me the solution. When I am using IAX, It is working fine always. What is the problem with SIP?Looking forward to your response.ThanksRegards, Chandra. See the all-new, redesigned Yahoo.com. Check it out. ___ --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] Please suggest me Best VoIP Service Provider
Hi Gbenga Great,I want to make calls to USA from INDIA using our "Asterisk". The sound quality should be good and able to make 4 Simulatanious calls at a time to USA. Do u have any unlimited plan to make calls to USA from INDIA using Asterisk? Please provide full details of your service. Looking forward to your response.ThanksRegards,Chandra.Gbenga Great [EMAIL PROTECTED] wrote: Hello chandra, What is your volume and target, we could provide you with USA route using your asteriks gbenga---Original Message--- From: Crazy Boy Date: 07/19/06 06:59:19 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Please suggest me Best VoIP Service Provider Hi Friends,I am Chandra from India. Currently, we are implementing Asterisk in our organization and using "Teliax" service to make calls to USA. Initially, Teliax service is very good. But, from 20 days, Teliax service is very poor and they are not responding for customer querys also. So, we want to change the VoIP Service provider. Our main aim is to make calls to USA from INDIA using "Asterisk". Can you please suggest me some best VoIP Service providers? Looking forward to your response. Thank you.Regards,Chandra. Groups are talking. We�re listening. Check out the handy changes to Yahoo! Groups. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users See the all-new, redesigned Yahoo.com. Check it out. ___ --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] Please suggest me Best VoIP Service Provider
Hi Ram,I am also located in Hyderabad only. Please give me your website. I will go through your website. Looking forward to your response. Thank you.Regards,Chandra.ram [EMAIL PROTECTED] wrote: Hi we are located in hyderabad (india) where are you located ? we do support DID incoming and out going we have veryresonable rates for USA and other Countries contact me with your Phone ram On 7/24/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Gbenga Great,I want to make calls to USA from INDIA using our "Asterisk". The sound quality should be good and able to make 4 Simulatanious calls at a time to USA. Do u have any unlimited plan to make calls to USA from INDIA using Asterisk? Please provide full details of your service. Looking forward to your response. ThanksRegards,Chandra. Gbenga Great [EMAIL PROTECTED] wrote:Hello chandra, What is your volume and target, we could provide you with USA route using your asteriks gbenga---Original Message--- From: Crazy Boy Date: 07/19/06 06:59:19 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Please suggest me Best VoIP Service Provider Hi Friends,I am Chandra from India. Currently, we are implementing Asterisk in our organization and using "Teliax" service to make calls to USA. Initially, Teliax service is very good. But, from 20 days, Teliax service is very poor and they are not responding for customer querys also. So, we want to change the VoIP Service provider. Our main aim is to make calls to USA from INDIA using "Asterisk". Can you please suggest me some best VoIP Service providers? Looking forward to your response. Thank you. Regards,Chandra. Groups are talking. We�re listening. Check out the handy changes to Yahoo! Groups.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users See the all-new, redesigned Yahoo.com. Check it out. ___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Music Unlimited - Access over 1 million songs. Try it free. ___ --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] How to connect XLite with another public IP?
Hi Friends,We have two internet connections (lines) from two Internet Service Providers in our office. So, we have two public IP addresses. We are using one connection for our LAN and to providing internet to our office staff. We are not using second connection. Now, we have installed "Asterisk" and using for International dialing with Second connection. Now, I have installed "XLite" softphone in our staff systems. I tried to connect our XLite with our Asterisk server. But, our XLite softphone is unable to connect with Asterisk server. I have given Asterisk server public IP in my XLite Domain field. But, it is not connecting and is giving an error i.e., " Registration error: 408 - Request timedout". I tried using firewall and without using firewall. Please tell me how to configure my XLite softphone to connect with my Asterisk server (With other public IP)?This is very urgent. Looking forward to your kind response. Thank you.Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
