Carrier problem

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Carrier problem

Postby bohan » Mon Nov 02, 2015 11:25 am

hello everyone! i have learnt a lot from this forum keep it up guys! now i have a problem.I installed vicibox 6.0.4 on a single server and everything went okay ,but i am getting a problem with connecting to the carrier. i cant make any call from the soft phone (eyebeam in this case) the carrier dont show in sip show registry 0 host found and yes i can ping the carrier from the server ...i dont know what i am doing wrong ..hope anyone can help me .. below are the specs
Vicibox 6.0.4 from .iso | Vicidial VERSION: 2.12-516a BUILD: 151022-1404| Asterisk 1.8.32.3| Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Xeon(R) CPU E31220 @ 3.10GHz Quad Core Ram 8gb HDD 250gb

carrier settings

register =>@x.x.x.x
[contabo]
disallow=all
allow=g729
port=5060
insecure=invite
type=friend
host=x.x.x.x
dtmfmode=rfc2833
trustrpid=yes
sendrpid=yes
canreinvite=no
qualify=no
context=custom-a2billing

global string: SIPTRUNK= SIP/contabo
exten => _39XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _39XXX.,2,Dial(${SIPTRUNK}/${EXTEN},60,tTo)
exten => _39XXX.,3,Hangup



server1*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.



server1*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
101/101 192.168.5.121 D N 36914 OK (100 ms)
contabo x.x.x.x N 5060 Unmonitored
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline]


what happen when i try to make a call
    server1:/usr/src # asterisk -r
    Asterisk 1.8.32.3-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
    Created by Mark Spencer
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    Connected to Asterisk 1.8.32.3-vici currently running on server1 (pid = 5917)
    Verbosity is at least 21
    [Nov 2 16:02:13] == Using SIP RTP CoS mark 5
    [Nov 2 16:02:13] -- Executing [39043746647@default:1] AGI("SIP/101-00000018", "agi://127.0.0.1:4577/call_log") in new stack
    [Nov 2 16:02:14] -- <SIP/101-00000018>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    [Nov 2 16:02:14] -- Executing [39043746647@default:2] Dial("SIP/101-00000018", "SIP/contabo/39043746647,60,tTo") in new stack
    [Nov 2 16:02:14] == Using SIP RTP CoS mark 5
    [Nov 2 16:02:14] -- Called SIP/contabo/39043746647
    [Nov 2 16:02:46] WARNING[13752]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 5a38b1813258519c722ef93a5294c856@X.Y.Z.W:5060 for seqno 102 (Critical Request) -- See wiki.asterisk /wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32000ms with no response
    [Nov 2 16:02:46] WARNING[13752]: chan_sip.c:3851 retrans_pkt: Hanging up call 5a38b1813258519c722ef93a5294c856@X.Y.Z.W:5060:5060 - no reply to our critical packet (see wiki.asterisk /wiki/display/AST/SIP+Retransmissions).
    [Nov 2 16:02:46] -- SIP/contabo-00000019 is circuit-busy
    [Nov 2 16:02:46] == Everyone is busy/congested at this time (1:0/1/0)
    [Nov 2 16:02:46] -- Executing [39043746647@default:3] Hangup("SIP/101-00000018", "") in new stack
    [Nov 2 16:02:46] == Spawn extension (default, 39043746647, 3) exited non-zero on 'SIP/101-00000018'
    [Nov 2 16:02:46] -- Executing [h@default:1] AGI("SIP/101-00000018", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CONGESTION----------") in new stack
    [Nov 2 16:02:46] -- <SIP/101-00000018>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Last edited by bohan on Tue Nov 03, 2015 10:18 am, edited 1 time in total.
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Re: Carrier problem

Postby nandotech » Mon Nov 02, 2015 5:52 pm

bohan wrote:global string: SIPTRUNK= SIP/contabo
exten => _39XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _39XXX.,2,Dial(${SIPTRUNK}/${EXTEN},60,tTo)
exten => _39XXX.,3,Hangup

server1*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
101/101 192.168.5.121 D N 36914 OK (100 ms)
contabo x.x.x.x N 5060 Unmonitored
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline]


Based on your "sip show peers" it looks like you might actually be attempting to connect contabo (just says "Unmonitored").

Based on your asterisk output, you are either not properly registering or need to have your IP whitelisted properly by your carrier (which more often than not, I have had to open a trouble ticket for).
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Re: Carrier problem

Postby williamconley » Mon Nov 02, 2015 5:55 pm

thanks for posting your system specs! :)

being able to ping does not mean that they will respond to a sip request. different ports.

if you have no entry for Carrier->Registration String, sip show registry will not show anything ... because you are not registering. Registration is usually not necessary. However, if you are NOT using registration, then the carrier must know who you are by knowing your IP address instead.

Two methods of authentication:

User/Pass (Registration) Authentication -> You are provided with a user/pass (username/secret, they can call it many things). SIP protocol is fairly straightforward in this arena, and the registration string is rather rigid in format. But if they did not give you a user/pass/domain/port to register ... then you must use:

IP Address Authentication -> You provide the carrier with your IP address. Usually in a web portal of some sort. Sometimes via email if they have no portal.

You can also usually use "qualify=1000" to find out if you are likely to be able to make a call. If qualify=1000 results in "UNREACHABLE", then port 5060 is closed to you.

Presently you appear to have "qualify=no", thus asterisk is not checking to see if the connection is valid at all. It's just sending the Invite packet off and "hoping for the best". But never getting a response. That's where the list of problems to check comes in:

Either there is a firewall problem (at your end or their end) or there is not a sip server at that IP address, or you are not allowed to contact their sip server (ie: your IP address is not allowed in their system).
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Re: Carrier problem

Postby bohan » Tue Nov 03, 2015 5:02 am

thanx for ur answers!yes i am not using a registration carrier.. and i am using the same carrier with an old server using goautodial 2.1 with the same carrier settings and it is working fine. so the other server is in the same network and has the same public ip adress so i think the ip is in whitelist and i have looked at open ports in yast->security setting and i can see port 5060 open there ..

i tried to change qualify=1000 and still not able to make a call ..thats the output i get when i make a call

== Using SIP RTP CoS mark 5
[Nov 3 10:56:20] -- Executing [39043746647@default:1] AGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 3 10:56:20] -- <SIP/101-00000000>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 3 10:56:20] -- Executing [39043746647@default:2] Dial("SIP/101-00000000", "SIP/contabo/39043746647,60,tTo") in new stack
[Nov 3 10:56:20] == Using SIP RTP CoS mark 5
[Nov 3 10:56:20] -- Called SIP/contabo/39043746647
[Nov 3 10:56:27] WARNING[5575]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 5e0e4811753a649c7d7923261de89d9d@X.Y.Z.W:5060:5060 for seqno 102 (Critical Request) -- See wiki.asterisk./wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Nov 3 10:56:27] WARNING[5575]: chan_sip.c:3851 retrans_pkt: Hanging up call 5e0e4811753a649c7d7923261de89d9d@X.Y.Z.W:5060:5060 - no reply to our critical packet (seewiki.asterisk./wiki/display/AST/SIP+Retransmissions).
[Nov 3 10:56:27] == Everyone is busy/congested at this time (1:0/0/1)
[Nov 3 10:56:27] -- Executing [39043746647@default:3] Hangup("SIP/101-00000000", "") in new stack
[Nov 3 10:56:27] == Spawn extension (default, 39043746647, 3) exited non-zero on 'SIP/101-00000000'
[Nov 3 10:56:27] -- Executing [h@default:1] AGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CHANUNAVAIL----------") in new stack
[Nov 3 10:56:27] -- <SIP/101-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Last edited by bohan on Tue Nov 03, 2015 10:21 am, edited 2 times in total.
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Re: Carrier problem

Postby bohan » Tue Nov 03, 2015 9:37 am

btw when i try to call manualy in eyebeam i dont hear any ring just at the end " the person u are calling is unavaible please try again" when i change the carrier setting to
exten => _39XXX.,2,Dial(${SIPTRUNK}/${EXTEN},60,tTor)
from
exten => _39XXX.,2,Dial(${SIPTRUNK}/${EXTEN},60,tTo)
every time i try to make a manual call throught eyebeam i can hear the ring but in every call it say " the person u are calling is unavaible please try again" and i know the person is avaible cuz i called them 1 min before

this is the output i get

[Nov 3 15:59:35] == Using SIP RTP CoS mark 5
[Nov 3 15:59:35] -- Executing [390439779059@default:1] AGI("SIP/101-0000004d", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 3 15:59:35] -- <SIP/101-0000004d>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 3 15:59:35] -- Executing [390439779059@default:2] Dial("SIP/101-0000004d", "SIP/contabo/390439779059,60,tTor") in new stack
[Nov 3 15:59:35] == Using SIP RTP CoS mark 5
[Nov 3 15:59:35] -- Called SIP/contabo/390439779059
[Nov 3 15:59:42] WARNING[5586]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 1e882dfb57fe3f0b783704715a22ec1d@X.Y.Z.W:5060:5060 for seqno 102 (Critical Request) -- See /wiki.asterisk./wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Nov 3 15:59:42] WARNING[5586]: chan_sip.c:3851 retrans_pkt: Hanging up call 1e882dfb57fe3f0b783704715a22ec1d@X.Y.Z.W:5060:5060 - no reply to our critical packet (see wiki.asterisk./wiki/display/AST/SIP+Retransmissions).
[Nov 3 15:59:42] == Everyone is busy/congested at this time (1:0/0/1)
[Nov 3 15:59:42] -- Executing [390439779059@default:3] Hangup("SIP/101-0000004d", "") in new stack
[Nov 3 15:59:42] == Spawn extension (default, 390439779059, 3) exited non-zero on 'SIP/101-0000004d'
[Nov 3 15:59:42] -- Executing [h@default:1] AGI("SIP/101-0000004d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CHANUNAVAIL----------") in new stack
[Nov 3 15:59:42] -- <SIP/101-0000004d>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... VAIL------ completed, returning 0
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Re: Carrier problem

Postby nandotech » Tue Nov 03, 2015 11:23 am

Once again..

1. You need to take the register string out
[Nov 3 15:59:42] WARNING[5586]: chan_sip.c:3851 retrans_pkt: Hanging up call 1e882dfb57fe3f0b783704715a22ec1d@X.Y.Z.W:5060:5060 - no reply to our critical packet


2. If you are supposed to be using IP authentication, you need to ensure that you are whitelisted with your carrier and/or open a ticket with them showing this output--they can usually help you get your configuration right. Or just allow your IP, which is probably the only problem.
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Re: Carrier problem

Postby bohan » Tue Nov 03, 2015 12:02 pm

nandotech wrote:Once again..

1. You need to take the register string out
[Nov 3 15:59:42] WARNING[5586]: chan_sip.c:3851 retrans_pkt: Hanging up call 1e882dfb57fe3f0b783704715a22ec1d@X.Y.Z.W:5060:5060 - no reply to our critical packet


2. If you are supposed to be using IP authentication, you need to ensure that you are whitelisted with your carrier and/or open a ticket with them showing this output--they can usually help you get your configuration right. Or just allow your IP, which is probably the only problem.


Yes i am usign IP authentication ..like i said before i know my ip is in white list because the other server on my network use the same carrier and it works perfectly
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Re: Carrier problem

Postby bohan » Thu Dec 10, 2015 5:58 am

i solved this by commenting the externip line in sip.conf
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Re: Carrier problem

Postby rrb555 » Wed Jan 06, 2016 9:12 pm

i have this uncommented. the value for my externip is my public IP.

Should we really comment or disable externip?
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