[asterisk-users] IAX softphone
Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soft phones.
Hi Ken, Can it be an IAX client? If so, I'd recommend KIAX. I used it once, both on Linux and Windows, and it worked for me. []s Ronaldo. On Thu, Jul 22, 2010 at 4:14 PM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if anyone has any favorites. Failing multi-platform, I'll stick with Twinkle on Linux, and gladly take suggestions for Windows -- OSS if possible, but payware is acceptable. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dundi x ENUM
Hi all, I've just being wondering if Dundi has the same purpose as ENUM. I don't know much (actually almost nothing) about these technologies. As far as I know they are a kind of DNS resolver used in the VoIP context. For example, user [EMAIL PROTECTED] has the extension namber 1001. This way nobody has to know the ronaldo's extension number. I'll appreciate if someone can clear my understanding about that? Thanks in advance. Ronaldo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Hi, I'm brazilian. By the way, Why such a question? See you. Ronaldo. Jozeph Brasil wrote: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunking using a different port
Hi all, Is it possible having a trunk using, for exemple, UDP port 4570 and all the other IAX (not trunk) connection using the standard UDP port 4569? Thanks. Ronaldo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX client USB phone
Hi all, Does anybody know any USB phone that I can use as an IAX Client? Thanks. Ronaldo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Path Optimization
Hi all, I was reading an IAX RFC, or a kind of, and it mentioned something about Call Path Optimization. Does Asterisk provide such a feature? Thanks. Ronaldo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STDERR in AGI
Hi all,
I just started programming using AGI and I have a simple question about
STDERR.
If I understood it right, all the messages sent to STDERR should be
shown in the Asterisk console, but using the following python code I
just can't see anything.
#!/usr/bin/python
#
# File: /var/lig/asterisk/agi-bin/agi-test.py
#
# Description: An AGI Script
#
import sys
env = {}
tests = 0
while True:
line = sys.stdin.readline().strip()
if line == '':
break
key,data = line.split(':')
if key[:4] != 'agi_':
sys.stderr.write(Did not work!\n)
sys.stderr.flush()
continue
key = key.strip()
data = data.strip()
if key != '':
env[key] = data
sys.stderr.write(AGI Environment Dump:\n)
for key in env.keys():
sys.stderr.write( -- %s = %s\n % (key,env[key]))
sys.stderr.flush()
##
This code comes from the book Asterisk: The future of the Internet and
it is being activated by an extension like that:
exten = 123,1,Answer()
exten = 123,2,AGI(agi-test.py)
Any help would be appreciated.
Ronaldo.
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[asterisk-users] AGI command
Hi all, Does anybody know why my asterisk doesn't have a show agi command? Do I have to load any module for it? Thanks Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
Hi Anthony, It doesn't make sense. This peer is an IAX peer. It was supposed to use UDP. Does Asterisk also use TCP for IAX? Thanks Ronaldo. Anthony Francis wrote: Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. ___ --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 T is for TCP, U would be UDP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk with dynamic IPs
Hi Jaswinder, That is what I did. The thing now is, when I set enable=yes in /etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the remote peer (to keep the trunk up). I've searched on the Internet but I couldn't find any documentation about how DNS update manager works for Asterisk. Do you have any? Ronaldo. Jaswinder Singh wrote: In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Peers show command
Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk with dynamic IPs
Hi Noah, First of all, thanks for your help. I just want to check if I understood. If a set the TTL for 10 seconds for host.no-ip.org and configure the parameter host as host=host.no-ip.org, Asterisk will try to find the IP address of host.no-ip.org each 10 seconds? That is it? Thanks again. Ronaldo. Noah Miller wrote: Hi Ronaldo - I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the host parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a iax2 reload in the CLI, asterisk tries to find the IP of the peer (based on its DNS name) and everything starts working again. This is the section for my trunk in one of my servers: [sometrunk] type=friend username=someusername secret=somesecret auth=plaintext host=host.no-ip.org context=incoming peercontext=incoming qualify=yes trunk=yes Is there any way to tell asterisk to try to find the peer's IP address if that peer is unreachable or each 10 minutes? I don't know if your DDNS provider would support this, but if you set the TTL value of your DNS hostnames to something very low, like 10 seconds, it would force your OS to keep finding the latest IP. - 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk with dynamic IPs
Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunk with dynamic IPs
Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the host parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a iax2 reload in the CLI, asterisk tries to find the IP of the peer (based on its DNS name) and everything starts working again. This is the section for my trunk in one of my servers: [sometrunk] type=friend username=someusername secret=somesecret auth=plaintext host=host.no-ip.org context=incoming peercontext=incoming qualify=yes trunk=yes Is there any way to tell asterisk to try to find the peer's IP address if that peer is unreachable or each 10 minutes? Thanks in advance. Ronaldo. ___ --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] X100P Clone
Hi all, I'm planning to buy a X100P clone and would like some feedback about this card. Does anyone already used this card? Does anyone recommend it ? or not? I'd appreciate any comments. Thanks. Ronaldo. ___ --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] The purpose of DUNDi
Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax to iax Reject Connection
Hi, Don't you have to configure the host option for each channel in iax.conf? Look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf Ronaldo. chawki hammoud wrote: Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. chan_iax2.c:5550 socket_read: Call rejected by : No authority found iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele disallow=all allow=gsm Any suggestions of why the connection is refused. I have no firewall. Thanks We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Were i make mistake
Hi Ardit, You set two session with the same name ([test]). Try changing those names to [test1] and [test2] Ronaldo. Ardit Saliu wrote: I’ve found some manuals and tried this to do : Sip.conf [test] type=friend username=test1 secret=test1 host=192.168.1.238 context=tutorial fromuser=SIP Phone callerid=101 nat=no canreinvite=yes dtfmode=info disallow=all allow=ulaw [test] type=friend username=test secret=test host=192.168.1.240 context=tutorial callerid=100 nat=no canreinvite=yes dtfmode=info disallow=all allow=ulaw Extensions.conf [tutorial] exten = 101,1,Dial(SIP/test1) exten = 100,1,Dial(SIP/test) allow restarted with sudo /etc/rc.d/init.d/asterisk restart I’m using Xlite 3.0 34025 In the phone shows “Registration error: 404 – Not found” What should I do now?? I just want to call from one phone to another 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 ___ --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] Problem with conferences, Vlada, Pancevo
Hi, I'm not sure, but MeetMe needs some timer module from zaptel project. Try read about timers for MeetMe application. Ronaldo. Vladimir Kovacevic wrote: Hi, I have problem with setting up a conferences. When I dial the reserved conference number from xlite the line gets hunged up and on a console there is a following message: WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe' for extension (internal, 1234, 3) exten = 1234,1,Answer() exten = 1234,4,MeetMe(1234|Md) exten = 1234,101,HangUp() meetme.conf: [general] [rooms] conf = 1234 What I did wrong? Thx, Vlada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi All, I'd like to thank everyone that answer my question about IAX Trunk. Now I have a working IAX trunking, I just need to tune it. Thank you. Ronaldo. Salvatore Giudice wrote: Yes of course. If you want to limit it, I think you have to set 'incominglimit' and/or 'outgoinglimit'. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, May 03, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? Thanks. Steve Kennedy wrote: On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Trunk
Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi Dean, Can you suggest me any documentation about using IAX trunking? Thank you. Ronaldo. Dean Collins wrote: Yes it is. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, 3 May 2007 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IAX Trunk Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? Thanks. Steve Kennedy wrote: On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) 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
[asterisk-users] SIP Proxy
Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router http://www.voip-info.org/wiki/view/SIP+Express+Router: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org http://www.voip-info.org/wiki/view/Vovida.org * sipX http://www.voip-info.org/wiki/view/sipX from Sipfoundry http://www.voip-info.org/wiki/view/SIPfoundry is a native SIP proxy but also a complete SIP PBX * OpenSER http://www.voip-info.org/wiki/view/OpenSER - scalable and robust SIP server with TLS support Can anyone suggest me something about these SIP Proxy? p.s) Is Asterisk a SIP Proxy? Regards ... Ronaldo. ___ --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] MeetMe Error
Hi, Check if your system has the /dev/files needed. I think some installation didn't do it automatically. Manolet Gmail wrote: 2007/4/18, Ronaldo [EMAIL PROTECTED]: Hi Manolet, You have to install zaptel in order to make MeetMe application to work. MeetMe needs a kind of timer device that is provided by zaptel package. Eventhough you don't have a zaptel card you need to install its package. Search for MeetMe application in http://www.voip-info.org/ and you will find documentation about how to do that. Good Luck. Ronaldo Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --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 but i have zaptel 1.4.1 installed... there is any special configuration or something? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Error
Hi Manolet, You have to install zaptel in order to make MeetMe application to work. MeetMe needs a kind of timer device that is provided by zaptel package. Eventhough you don't have a zaptel card you need to install its package. Search for MeetMe application in http://www.voip-info.org/ and you will find documentation about how to do that. Good Luck. Ronaldo Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Applications
Ok Eric, Thank you again. Ronaldo On 4/14/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Ronaldo Zacarias Afonso wrote: Hi all, I was trying to set up a conference room using the MeetMe application and my asterisk is telling me that there is no MeetMe application available for the extension I've dialed. [Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for extension (internal, 600, 1) So, I issued the command core show applications and it didn't show me a MeetMe application. I'd like to know how I can install that application or what I have to do to make it available for use. Thanks. MeetMe requires Zaptel to be installed. Install Zaptel, then rebuild and install Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing Applications
Hi all, I was trying to set up a conference room using the MeetMe application and my asterisk is telling me that there is no MeetMe application available for the extension I've dialed. [Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for extension (internal, 600, 1) So, I issued the command core show applications and it didn't show me a MeetMe application. I'd like to know how I can install that application or what I have to do to make it available for use. Thanks. Ronaldo. ___ --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: number to names
OK Yuan, What I wanted to know is if the extension I've created is right. exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) Will my asterisk bridge a SIP phone that dialed 101 to the SIP user: [EMAIL PROTECTED] Do I need some think more in order for it to work? Do you have or know any documentation that explains me that? Regards Ronaldo. On 4/13/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Ronaldo Zacarias Afonso [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 11:54:51 -0300 Hi all, Is it possible to configure an extension number to dial a sip address? Nothing prevents you from doing this. Yuan Liu For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP: number to names
Hi all, Is it possible to configure an extension number to dial a sip address? For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hard phone configuration
Hi, It's really a simple question! I've just started playing with asterisk too, and I think what you want could be found in the 4th chapter of Asterisk: The Future of the Internet. It's a open book you can download from http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11. I hope it'd helped you. Ronaldo. On 4/12/07, Ilya Vishnyakov [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: How do I make Asterisk communicate with my Grandstream hard phone? Thank you in advance. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHovpUZGmaUWxLn8RAn9UAJ94exp6gs2PBWpMDiiNA69Mt78jhgCfYy71 eOq4eOuYi2uDpve+8YM2fp4= =+Jt7 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Hi Stephen, I'm using Grandstream and I think is a nice phone, but its the only one that I've tried. I bought it to learn voip/asterisk. Just my 2 cents. Good luck. Ronaldo. On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) -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 ___ --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] Require only GSM Codec
Hi Sanjay, I'm not sure about that, but I think you can configure it in, for example, /etc/asterisk/sip.conf. There is an option that you configure for each channel like: only=gsm It instructs the sip protocol, that only gsm codec must be used. I hope it has helped you. Regards, Ronaldo. On 4/3/07, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. 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 ___ --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] Linking incoming calls
Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo. ___ --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] question about DISA
Lists, Hi, good day, i was being task to create a DISA access for internal purpose of the company, i'm having a problem to work with it with authentication, but i think it's really a straight forward thing to do, can someone enlight me on this. thanks sample code snippet exten = 5,Goto(inward,s,1) [inward] exten = s,1,Disa(1234|outgoing) ; DISA apps supposed to ask me a password but it's not instead it's drop me immedietly to a dial tone exten = s,2,Hangup My Workaround. exten = s,1,Authenticate(1234) exten = s,2,Disa(no-password|outgoing) Thanks Ronald ___ --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] Integration with SIEMENS HIPATH PBX
Hi, I would like to know if Asterisk is able to be integrated with a Siemens HIPATH PBX by VoIP or other ways. Best regards, Ronaldo S. Pereira PRI Telemática.
