jonas kellens
2009-Apr-13 16:18 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr = 0.0.0.0
[BT201]
type=friend
context=intern
host=192.168.4.210
secret=testpaswoord
[GXP1200]
type=friend
context=intern
host=192.168.4.211
secret=testpaswoord
extensions.conf :
[root at asterisk asterisk]# cat extensions.conf
[intern]
exten => 210,1,Dial(SIP/BT201)
exten => 211,1,Dial(SIP/GXP1200)
Asterisk CLI shows me :
asterisk*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
== Parsing '/etc/asterisk/sip_notify.conf': Found
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
GXP1200 192.168.4.211 5060
Unmonitored
BT201 192.168.4.210 5060
Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]
asterisk*CLI> dialplan show intern
[ Context 'intern' created by 'pbx_config' ]
'210' => 1. Dial(SIP/BT201)
[pbx_config]
'211' => 1. Dial(SIP/GXP1200)
[pbx_config]
I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.
I would love to have your feedback on this. Where could this problem be
situated ?
I notice (on the Asterisk CLI) that my SIP-phones do not register. They
have a fixed IP and there account information is set via the web
interface.
Greetingz,
Jonas.
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Michael van der Stoop
2009-Apr-13 16:28 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
jonas kellens wrote:> Hi there, > > this is the first time that I'm building an Asterisk-server. > > I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. > Zaptel is for later, when configuring the POTS-line. Now first > internal communication with SIP. > > Thought it would go easier... > > I have 2 Grandstream IP-phones : BT-201 and GXP-1200. > > These are my settings : > > sip.conf : > /[root at asterisk asterisk]# cat sip.conf/ > /[general]/ > /bindport=5060/ > /bindaddr = 0.0.0.0/ > > /[BT201]/ > /type=friend/ > /context=intern/ > /host=192.168.4.210/ > /secret=testpaswoord/ > > /[GXP1200]/ > /type=friend/ > /context=intern/ > /host=192.168.4.211/ > /secret=testpaswoord/ > extensions.conf : > /[root at asterisk asterisk]# cat extensions.conf/ > /[intern]/ > /exten => 210,1,Dial(SIP/BT201)/ > /exten => 211,1,Dial(SIP/GXP1200)/ > Asterisk CLI shows me : > /asterisk*CLI> sip reload/ > /Reloading SIP/ > / == Parsing '/etc/asterisk/sip.conf': Found/ > / == Parsing '/etc/asterisk/users.conf': Found/ > / == Parsing '/etc/asterisk/sip_notify.conf': Found/ > /asterisk*CLI> sip show peers/ > /Name/username Host Dyn Nat ACL Port > Status / > /GXP1200 192.168.4.211 5060 > Unmonitored / > /BT201 192.168.4.210 5060 > Unmonitored / > /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 > offline]/ > > /asterisk*CLI> dialplan show intern/ > /[ Context 'intern' created by 'pbx_config' ]/ > / '210' => 1. Dial(SIP/BT201) > [pbx_config]/ > / '211' => 1. Dial(SIP/GXP1200) > [pbx_config]/ > > I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem > be situated ? > > I notice (on the Asterisk CLI) that my SIP-phones do not register. > They have a fixed IP and there account information is set via the web > interface. > > Greetingz, > Jonas. > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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-usersI had the same issue. I set the hosts to dynamic and and explicitly set their IP's via a dhcp server using their MAC addresses. The phones registered and all is well. Regards, Mike
Anthony Plack
2009-Apr-13 16:34 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
> bindaddr = 0.0.0.0 >I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack
"ROQUÉ, Francisco Emiliano"
2009-Apr-13 16:55 UTC
[asterisk-users] ***SPAM*** Re: Asterisk-beginner : cannot make phonecalls using Asterisk
In your sip.conf or sip_nat.conf (for elastix) set the variables: externhostexternipdomainexternrefreshlocalnet Regards Francisco Anthony Plack wrote:>> bindaddr = 0.0.0.0 >> >> > > I would set this to the ethernet interface IP address, I believe this may be your issue. > > Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. > > Tony Plack > > > > _______________________________________________ > -- 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 > > >
Steve Edwards
2009-Apr-13 16:59 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
On Mon, 13 Apr 2009, Anthony Plack wrote:>> bindaddr = 0.0.0.0 > > I would set this to the ethernet interface IP address, I believe this > may be your issue.Binding to 0.0.0.0 means "listen to all IP addresses on the box." It is not the issue. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
James A. Shigley
2009-Apr-13 16:59 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
What do you see when you run asterisk ?r and dial 210 or 211 from one of the phones James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by "reply to sender only" message and destroy all electronic and hard copies of the communication, including attachments. "Common sense is the collection of prejudices acquired by age eighteen." -- Albert Einstein "Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy." -- Albert Einstein "I know a little of everything, but a lot of nothing" From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [root at asterisk asterisk]# cat extensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status GXP1200 192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI> dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' => 1. Dial(SIP/BT201) [pbx_config] '211' => 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090413/61ceb8cb/attachment.htm
Barry L. Kline
2009-Apr-13 18:04 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 jonas kellens wrote:> I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : > > /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/ > /Verbosity is at least 5/ > /asterisk*CLI> / > > Nothing is displayed... it stays that way... > > Jonas.Is there a "Send" button on that phone? It sounds to me as though the phone is still waiting for more digits. Barry -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ437BCFu3bIiwtTARAs73AJ9spwpr7ULu6VyimPPoDIPnzFK6JQCbBEDO bQ0m2dROkUEkdtwCHtbHTBI=4Zmk -----END PGP SIGNATURE-----
Doug Lytle
2009-Apr-13 19:24 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
jonas kellens wrote:> Hi there, > > I notice (on the Asterisk CLI) that my SIP-phones do not register. > They have a fixed IP and there account information isIf your phones don't register, then your not going to be able to make a call. The Grandstream phones have a web interface (At least if memory serves correctly) and you'll need to tell the phones: server ip address username/extension to use password of that extension. Once they've registered, things will get a lot easier. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Tzafrir Cohen
2009-Apr-13 20:11 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
Hi On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:> I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem be > situated ?Your basic mistake at troubleshooting this is trying to test two things at the same time. Let's test them separately. 1. A call from Asterisk to the phones: In the Asterisk CLI: originate SIP/BT201 application playback demo-instruct And the other one: originate SIP/GXP1200 application playback demo-instruct Alternatively, use the echo-test aplication: originate SIP/BT201 application echo 2. Next, test calling from the phones to Asterisk. Add those two extensions to [intern] exten => 250,1,Answer exten => 250,n,Playback(demo-instruct) exten => 250,n,Hangup exten => 251,1,Answer exten => 251,1,Echo exten => 251,1,Hangup Make sure you reload for that to take effect, and then try dialing 250 or 251. Another useful tools: 'sip debug'. It tends to generate a very noisy output that is normally not readable for mere mortals. However it does indicate that "something is happening". If you call from a remote SIP phone and there's nothing on the SIP debug, the problem is probably with the settings of the phone, as it is not getting to you. Last and not least: a sanity check as you "see nothing": what is the output of: 'logger show channels' ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Danny Nicholas
2009-Apr-14 15:12 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
Put register=yes in the BT201 and GXP1200 contexts of sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [root at asterisk asterisk]# cat extensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status GXP1200 192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI> dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' => 1. Dial(SIP/BT201) [pbx_config] '211' => 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090414/63545f2c/attachment.htm
Casey Boone
2009-Apr-14 15:36 UTC
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
just for a test, run "service iptables stop" as root on the asterisk server and then reboot your phones. after that, try again and see if the phones are making communications with asterisk. you can turn the firewall back on with "service iptables start" jonas kellens wrote:> Hi there, > > this is the first time that I'm building an Asterisk-server. > > I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. > Zaptel is for later, when configuring the POTS-line. Now first internal > communication with SIP. > > Thought it would go easier... > > I have 2 Grandstream IP-phones : BT-201 and GXP-1200. > > These are my settings : > > sip.conf : > /[root at asterisk asterisk]# cat sip.conf/ > /[general]/ > /bindport=5060/ > /bindaddr = 0.0.0.0/ > > /[BT201]/ > /type=friend/ > /context=intern/ > /host=192.168.4.210/ > /secret=testpaswoord/ > > /[GXP1200]/ > /type=friend/ > /context=intern/ > /host=192.168.4.211/ > /secret=testpaswoord/ > extensions.conf : > /[root at asterisk asterisk]# cat extensions.conf/ > /[intern]/ > /exten => 210,1,Dial(SIP/BT201)/ > /exten => 211,1,Dial(SIP/GXP1200)/ > Asterisk CLI shows me : > /asterisk*CLI> sip reload/ > /Reloading SIP/ > / == Parsing '/etc/asterisk/sip.conf': Found/ > / == Parsing '/etc/asterisk/users.conf': Found/ > / == Parsing '/etc/asterisk/sip_notify.conf': Found/ > /asterisk*CLI> sip show peers/ > /Name/username Host Dyn Nat ACL Port Status > / > /GXP1200 192.168.4.211 5060 > Unmonitored / > /BT201 192.168.4.210 5060 > Unmonitored / > /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 > offline]/ > > /asterisk*CLI> dialplan show intern/ > /[ Context 'intern' created by 'pbx_config' ]/ > / '210' => 1. Dial(SIP/BT201) > [pbx_config]/ > / '211' => 1. Dial(SIP/GXP1200) > [pbx_config]/ > > I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem be > situated ? > > I notice (on the Asterisk CLI) that my SIP-phones do not register. They > have a fixed IP and there account information is set via the web interface. > > Greetingz, > Jonas. > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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
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