Basic Friend->Friend, call hangs


#1

So I’m fairly new to Asterisk. I’m trying to setup basic SIP phone->SIP phone calls. I just want my user 100 to be able to dial user 101, and it will give me the waiting call sound, and ring on 101. So far I’m not getting good results. I just have them registering, that at least works fine.

This is installed on Gentoo Linux, asterisk version “asterisk-1.8.5.0-r3”.

sip.conf

[code][general]
port = 5060
bindaddr = 0.0.0.0
context=default

[100]
type=friend
callerid=My Name <100>
secret=pass123
host=dynamic
mailbox=100@default
dtmfmode=inband

[101]
type=friend
callerid=Other Name <101>
secret=pass123
host=dynamic
mailbox=101@default
dtmfmode=inband[/code]

extensions.conf

[code][default]

exten => 100,1,Dial(SIP/100,20)
exten => 100,2,VoiceMail(100,u)

exten => 101,1,Dial(SIP/101,20)
exten => 101,2,VoiceMail(101,u)

exten => 199,1,VoiceMailMain(${CALLERID(num)},s)[/code]

EDIT: Here’s what is happening in the log “/var/log/asterisk/messages” when I call from 100 to 101.

[Oct 14 07:34:12] NOTICE[19177] cdr.c: CDR simple logging enabled. [Oct 14 07:34:13] NOTICE[19177] loader.c: 184 modules will be loaded. [Oct 14 07:34:13] NOTICE[19177] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Oct 14 07:34:13] WARNING[19177] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). [Oct 14 07:34:13] NOTICE[19177] config.c: Registered Config Engine mysql [Oct 14 07:34:13] NOTICE[19177] chan_skinny.c: Configuring skinny from skinny.conf [Oct 14 07:34:13] NOTICE[19177] pbx_ael.c: Starting AEL load process. [Oct 14 07:34:13] NOTICE[19177] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load [Oct 14 07:35:03] WARNING[19325] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Oct 14 07:35:04] WARNING[19325] app_voicemail.c: No entry in voicemail config file for '101'

I am able to register as both 100 and 101. But when I call from 100 to 101 or vice versa, it just says “Dialing” and I never hear any call tones and 101 never rings. Ideas?

Thanks in advance!


#2

Please provide the output after using sip set debug on.

It’s much easier when one sees what is actually happening than to have to guess.


#3

I added what’s happening in the log to the top post.


#4

You need at least verbose 3. Also I did ask for the SIP debugging output.

However, given the dialplan and the error message, I don’t believe the device you were trying to call is actually registered.

Maybe you think it is registered because you can call in, but that really just means that you are defaulting to allowguest=yes and bypassing the specific sip.conf sections.

This should cause a failover to the voicemail, so I still think we need more debugging output.


#5

[quote=“david55”]You need at least verbose 3. Also I did ask for the SIP debugging output.

However, given the dialplan and the error message, I don’t believe the device you were trying to call is actually registered.

Maybe you think it is registered because you can call in, but that really just means that you are defaulting to allowguest=yes and bypassing the specific sip.conf sections.

This should cause a failover to the voicemail, so I still think we need more debugging output.[/quote]

How do I get the SIP debugging output? Also, I added “allowguest=no” to my sip.conf.

This is what happened when I set startup opts with -vvvvv

> Saved useragent "Sipdroid/2.3 beta/Nexus One" for peer 100 == Using SIP RTP CoS mark 5 [Oct 14 08:03:58] NOTICE[23111]: chan_sip.c:21977 handle_request_invite: Call from '100' (67.159.153.121:45878) to extension 'asterisk' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 -- Executing [101@default:1] Dial("SIP/100-00000000", "SIP/101,20") in new stack [Oct 14 08:04:04] WARNING[23146]: app_dial.c:2196 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [101@default:2] VoiceMail("SIP/100-00000000", "101,u") in new stack [Oct 14 08:04:05] WARNING[23146]: app_voicemail.c:5602 leave_voicemail: No entry in voicemail config file for '101' -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [101@default:1] Dial("SIP/100-00000001", "SIP/101,20") in new stack [Oct 14 08:04:26] WARNING[23195]: app_dial.c:2196 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [101@default:2] VoiceMail("SIP/100-00000001", "101,u") in new stack [Oct 14 08:04:26] WARNING[23195]: app_voicemail.c:5602 leave_voicemail: No entry in voicemail config file for '101' -- Auto fallthrough, channel 'SIP/100-00000001' status is 'CHANUNAVAIL' [Oct 14 08:04:30] WARNING[23111]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 336252631582@192.168.1.101 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response


#6

Ok, when I ran the server locally with no firewalls I was able to get the call to go through, my phone 101 rings now, and I can pick it up and establish a call.

I only hear audio from 101->100 though, and on 101 I cant hear 100. Why is this? There are no firewalls, it’s a local network.

My Sipdroid client also keeps saying “Registration Failed: 405 method not allowed” after a little time being registered.


#7

sip set debug on


#8

On my local network I am now able to make/receive calls. The Ekiga client on Linux is so buggy, it caused most of the problems.

Thanks for trying to help.