Error Originating call from Extension

mikeosake · October 28, 2013, 12:03pm

Hi,

when i attempt to originate a call using a call file, see below:

Channel: SIP/ICNGw/2347061140114
Context: feelcall
Extension: 500
CallerID: 014403
Archive: yes

context:

[freecall]
exten => s,1,Answer(500)
exten => s,2,Playback(hello-world)
exten => s,3,Dial(SIP/voxbeam/00111032348098009386);
exten => s,4,Answer(500);
exten => s,5,Playback(hello-world)
exten => t,hangup();

i get the following error:
– Attempting call on SIP/ICNGw/2347061140114 for 500@feelcall:1 (Retry 1)
== Using SIP RTP CoS mark 5
> Channel SIP/ICNGw-0000076d was answered
[Oct 28 12:47:32] WARNING[25658][C-000007bd]: pbx.c:6390 __ast_pbx_run: Channel ‘SIP/ICNGw-0000076d’ sent to invalid extension but no invalid handler: context,exten,priority=feelcall,500,1
[Oct 28 12:47:32] NOTICE[25658]: pbx_spool.c:402 attempt_thread: Call completed to SIP/ICNGw/2347061140114

I runs the same config on another box and it works, although I don’t get any audio, as in both parties cannot hear themselves.

what am i doing wrong?

I want to originate 2 calls and join them so they can talk to them selves.

Regards,
Michael

mikeosake · October 28, 2013, 12:27pm

Hi,

I saw able to resolve this, was just calling the wrong extension, now the call is stuck on:

SIP/ICNGw-00000826 answered SIP/ICNGw-00000825
    -- Locally bridging SIP/ICNGw-00000825 and SIP/ICNGw-00000826

I hear nothing on both sides, what am I doing wrong?

Michael

david55 · October 28, 2013, 12:34pm

Lack of audio is normally a firewall (or NAT) problem.

mikeosake · October 28, 2013, 12:43pm

Server has no firewall.

Incoming calls are working well
Originated calls are working well
but this issue is just on Local bridge or 2 sip trunk calls.

do you have any samples on how to place call A, then dial call b and join/bridge both calls?

Regards,
Michael

david55 · October 28, 2013, 12:58pm

I don’t see how that would help, and it would be putting you into areas of the Asterisk code that are less likely to be well exerised and tested. Originate is fairly well exercised as it is heavily used to generate junk phone calls, and the outgoing leg is set up with the, very standard, Dial application.

You need to dump the SDP (sip. set debug on) and try and work out why the RTP isn’t getting through.

Also, unless you explicitly disable it, there will be a firewall on the machine running Asterisk.

Note that having two calls to Answer in the same basic block in the dialplan doesn’t make sense.

Actually, for Originate, the A leg is alreay answered by the time the extension is run, so neither of the calls to Answer do anything that Wait wouldn’t do.

mikeosake · October 28, 2013, 2:01pm

What do you suggest would be the best way to achieve this?
Party A is called,
Party A answers,
Party B is called
Party B answers then both calls are joined or bridged.

what your suggestions?

[/code]
[freecall]
exten => s,1,Answer(500)
exten => s,n,Playback(hello-world)
exten => s,n,Dial(SIP/ICNGw/23480980093xx);
[/code]

call is originated to Party A with this extension in the Call file

any assistance will be greatly appreciated.

Michael

david55 · October 28, 2013, 2:07pm

[freecall]
exten s,1,Dial(B)

where B is the technology and resource name for B.

Note Context needs to be freecall, not feelcall and Extension needs to be s, not 500.

mikeosake · October 28, 2013, 2:47pm

Hi,

please view the SIP DEBUG BELOW, i really don’t know how to read it:

v=0
o=root 2106000125 2106000125 IN IP4 208.109.198.60
s=Asterisk PBX 1.8.23.1
c=IN IP4 208.109.198.60
t=0 0
m=audio 14062 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/ICNGw/2348098009386
Retransmitting #1 (NAT) to 196.216.253.151:5060:
INVITE sip:2348098009386@196.216.253.151:5060 SIP/2.0
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK592f1acd;rport
Max-Forwards: 70
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>
Contact: <sip:014405103@208.109.198.60:5060>
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.1
Date: Mon, 28 Oct 2013 14:39:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 2106000125 2106000125 IN IP4 208.109.198.60
s=Asterisk PBX 1.8.23.1
c=IN IP4 208.109.198.60
t=0 0
m=audio 14062 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:196.216.253.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK592f1acd;rport=5060
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>;tag=d165f4872d449781
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 INVITE
Contact: <sip:SIPLOAD@196.216.253.151>
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER, PRACK, OPTIONS, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:196.216.253.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK592f1acd;rport=5060
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>;tag=d165f4872d449781
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 INVITE
Contact: <sip:SIPLOAD@196.216.253.151>
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER, PRACK, OPTIONS, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:196.216.253.151:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK592f1acd;rport=5060
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>;tag=d165f4872d449781
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 INVITE
Contact: <sip:SIPLOAD@196.216.253.151>
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER, PRACK, OPTIONS, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 194

v=0
o=SIPLOAD 123456 654321 IN IP4 196.216.253.151
s=-
c=IN IP4 196.216.253.156
t=0 0
m=audio 12130 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (10 headers 9 lines) ---
list_route: hop: <sip:SIPLOAD@196.216.253.151>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 196.216.253.156:12130
    -- SIP/ICNGw-00333aaf is making progress passing it to SIP/ICNGw-00333aae

<--- SIP read from UDP:196.216.253.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK592f1acd;rport=5060
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>;tag=d165f4872d449781
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 INVITE
Contact: <sip:SIPLOAD@196.216.253.151>
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER, PRACK, OPTIONS, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 194

v=0
o=SIPLOAD 123456 654321 IN IP4 196.216.253.151
s=-
c=IN IP4 196.216.253.156
t=0 0
m=audio 12130 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (10 headers 9 lines) ---
list_route: hop: <sip:SIPLOAD@196.216.253.151>
set_destination: Parsing <sip:SIPLOAD@196.216.253.151> for address/port to send to
set_destination: set destination to 196.216.253.151:5060
Transmitting (NAT) to 196.216.253.151:5060:
ACK sip:SIPLOAD@196.216.253.151 SIP/2.0
Via: SIP/2.0/UDP 208.109.198.60:5060;branch=z9hG4bK054030cc;rport
Max-Forwards: 70
From: "014405103" <sip:014405103@208.109.198.60>;tag=as55bdf6d1
To: <sip:2348098009386@196.216.253.151:5060>;tag=d165f4872d449781
Contact: <sip:014405103@208.109.198.60:5060>
Call-ID: 1dfe6f1a0e4343dc12f3a86e4feccee0@208.109.198.60:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.1
Content-Length: 0


---
    -- SIP/ICNGw-00333aaf answered SIP/ICNGw-00333aae
    -- Locally bridging SIP/ICNGw-00333aae and SIP/ICNGw-00333aaf

you can see it stops at Locally bridging … i have stopped the iptables service, so no firewall any where around.

Michael

david55 · October 28, 2013, 3:45pm

It’s incomplete. It only shows the outgoing setup ,and starts on a retry of the invite.

All the addresses are public and you see to have negotiated G.711 mu-Law codecs on that leg.

There is early media.

The remote media IP address is not the same as their SIP adddress.

Provided nothing is blocking IP4 196.216.253.156:12130 <> 208.109.198.60:14662 UDP, in either direction, the outgoing leg should be OK.

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Error Originating call from Extension

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