[asterisk-users] Fwd: Asterisk With Cisco Voice Router

Sat May 16 07:46:51 CDT 2009

Thanks Steve for this tip.

I have insecure=very is not yet deprecated. I have added it but still no good.

I personally think the problem could be with the codecs. Any ideas?

I have attached some debug info.

Regards,
Tim

On Sat, May 16, 2009 at 3:25 PM, Steve Howes <steve at geekinter.net> wrote:
>
> On 16 May 2009, at 12:46, Timothy Smith wrote:
>> <blah>
>>
>> Has anyone had the above set up working successfully? Attached are
>> some confs.
>>
>> Thanks a lot for your assistance.
>
> Check about the sip.conf 'insecure' option. I have had to use it in
> the past for similar stuff. I think it was 'insecure=very' but that
> might be deprecated by now..
>
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-------------- next part --------------
VG2#
VG2#
VG2#
VG2#
VG2#
May 16 12:41:40.237: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8  callref = 0x0C73
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98392
                Exclusive, Channel 18
        Calling Party Number i = 0x2183, '730230199'
                Plan:ISDN, Type:National
        Called Party Number i = 0xA1, '790792888'
                Plan:ISDN, Type:National
May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_INCOMING
May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x33B3, Guid = BCBB464BB098
VG2#
May 16 12:41:40.241: fb_get_reject_cause_code: ERROR cause_code NULL

May 16 12:41:40.245: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 12:41:40.245: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8C73
        Channel ID i = 0xA98392
                Exclusive, Channel 18
May 16 12:41:40.353: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 12:41:40.353: ISDN Se0/0/1:15 Q931: TX -> ALERTING pd = 8  callref = 0x8C73
VG2#
May 16 12:41:46.697: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret
May 16 12:41:46.697: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD
May 16 12:41:46.697: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8  callref = 0x8C73
May 16 12:41:46.713: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8  callref = 0x0C73
May 16 12:41:46.713: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_PROGRESS
OULKLAVG2#
May 16 12:41:46.713: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A
VG2#
May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x0C73
        Cause i = 0x8490 - Normal call clearing
May 16 12:41:52.373: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_DISC
May 16 12:41:52.373: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A
May 16 12:41:52.373: %ISDN-6-DISCONNECT: Interface Serial0/0/1:17  disconnected from 730230199 , call lasted 5 seconds
VG2#
May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8  callref = 0x8C73
May 16 12:41:52.385: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x0C73
May 16 12:41:52.385: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_CLEARED
-------------- next part --------------
From: "730230199" <sip:730230199 at 172.19.3.150>;tag=as5f114784
To: <sip:100 at 172.19.4.102:32544;rinstance=e6a140ee2d1dee0f>;tag=ae700477
Contact: <sip:730230199 at 172.19.3.150>
Call-ID: 7beff1bd661329c643aa69ec43628adc at 172.19.3.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.9
Content-Length: 0

---
    -- SIP/100-00820520 answered SIP/172.17.3.248-007fc920
Audio is at 172.19.3.150 port 13312
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.17.3.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK501204;received=172.17.3.248
From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA
To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9
Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:730232888 at 172.19.3.150>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 544232458 544232458 IN IP4 172.19.3.150
s=Asterisk PBX 1.6.0.9
c=IN IP4 172.19.3.150
t=0 0
m=audio 13312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/172.17.3.248-007fc920 and SIP/100-00820520
cs-intranet*CLI>
<--- SIP read from UDP://172.17.3.248:62582 --->
ACK sip:730232888 at 172.19.3.150:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK511A76
From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA
To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9
Date: Sat, 16 May 2009 12:38:27 GMT
Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
cs-intranet*CLI>