[asterisk-users] Calling '**1' through Asterisk
Rizwan Hisham
rizwanhasham at gmail.com
Thu May 29 06:15:15 CDT 2008
seems like asterisk could not find the extension number "**1". does your
context "from-internal" have this extensions? if not then create one or
include the context which contains the extension **1 in from-internal and
then try.
On Thu, May 29, 2008 at 1:10 AM, Henrik Ostergaard Madsen <
Henrik at ostergaard.net> wrote:
> Thanks for the reply.
>
> Good point with the features.conf. But I do not have any features.conf
> which conflict with **1 - and it is **2, **3 etc as well. Anyway, **10
> should be
> tricked by features as well, which it does not. And features only works on
> a
> bridged call, and this does not even get that far..
>
> The sip set debug peer <> did get me some extra output, but I am not able
> to get any sense from it. This is what came out (Asterisk is on
> 192.168.2.1
> AND 192.168.27.7 and the vopiphone is on 192.168.7.98 and has the
> username 018):
>
> <--- SIP read from 192.168.27.98:5060 --->
> INVITE sip:**1 at 192.168.2.1 SIP/2.0
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
> 58c7d35-4a11b393
> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER,
> SUBSCRIBE, NOTIFY, MESSAGE, INFO
> User-Agent: ATMEL_UA v0.0.25-alpha
> Max-Forwards: 70
> Contact: <sip:018 at 192.168.27.98:5060>
> Content-Type: application/sdp
> Content-Length: 334
>
> v=0
> o=018 12119074 77112119074177 IN IP4 192.168.27.98
> s=audio
> c=IN IP4 192.168.27.98
> t=0 0
> m=audio 16426 RTP/AVP 0 8 18 4 97 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11,16
> a=ptime:30
>
> <------------->
> --- (12 headers 15 lines) ---
> Sending to 192.168.27.98 : 5060 (NAT)
> Using INVITE request as basis request - 10b2f2f0-621ba8c0-13c4-40030-
> 16ba5-2f6d4897-16ba5
>
> <--- Reliably Transmitting (no NAT) to 192.168.27.98:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.27.98:5060;branch=z9hG4bK-16ba5-58c7d35-
> 4a11b393;received=192.168.27.98;rport=5060
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>;tag=as3903248a
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="ostergaard.net",
> nonce="19ae9086"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '10b2f2f0-621ba8c0-13c4-40030-
> 16ba5-2f6d4897-16ba5' in 32000 ms (Method: INVITE)
> Found user '018'
> Ildvaeg*CLI>
> <--- SIP read from 192.168.27.98:5060 --->
> ACK sip:**1 at 192.168.2.1 SIP/2.0
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>;tag=as3903248a
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 1 ACK
> Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
> 58c7d35-4a11b393
> Max-Forwards: 70
> Contact: <sip:018 at 192.168.27.98:5060>
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> Ildvaeg*CLI>
> <--- SIP read from 192.168.27.98:5060 --->
> INVITE sip:**1 at 192.168.2.1 SIP/2.0
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 2 INVITE
> Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
> 58c7d53-6bede6e9
> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER,
> SUBSCRIBE, NOTIFY, MESSAGE, INFO
> User-Agent: ATMEL_UA v0.0.25-alpha
> Max-Forwards: 70
> Contact: <sip:018 at 192.168.27.98:5060>
> Proxy-Authorization: Digest
> username="018",realm="ostergaard.net",nonce="19ae9086",uri="sip:**1 at 1
> 92.168.2.1",response="512fdfcf3ad644a79d92e4037679eee9",algorithm=M
> D5
> Content-Type: application/sdp
> Content-Length: 334
>
> v=0
> o=018 12119074 77112119074177 IN IP4 192.168.27.98
> s=audio
> c=IN IP4 192.168.27.98
> t=0 0
> m=audio 16426 RTP/AVP 0 8 18 4 97 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11,16
> a=ptime:30
>
> <------------->
> --- (13 headers 15 lines) ---
> Sending to 192.168.27.98 : 5060 (NAT)
> Using INVITE request as basis request - 10b2f2f0-621ba8c0-13c4-40030-
> 16ba5-2f6d4897-16ba5
> Found user '018'
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 4
> Found RTP audio format 97
> Found RTP audio format 101
> Peer audio RTP is at port 192.168.27.98:16426
> Found description format PCMU for ID 0
> Found description format PCMA for ID 8
> Found description format G729 for ID 18
> Found description format G723 for ID 4
> Found description format iLBC for ID 97
> Found description format telephone-event for ID 101
> Capabilities: us - 0x4 (ulaw), peer - audio=0x50d
> (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-
> event), combined - 0x0 (nothing)
> Peer audio RTP is at port 192.168.27.98:16426
> Looking for **1 in from-internal (domain 192.168.2.1)
>
> <--- Reliably Transmitting (no NAT) to 192.168.27.98:5060 --->
> SIP/2.0 484 Address Incomplete
> Via: SIP/2.0/UDP 192.168.27.98:5060;branch=z9hG4bK-16ba5-58c7d53-
> 6bede6e9;received=192.168.27.98;rport=5060
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>;tag=as3903248a
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '10b2f2f0-621ba8c0-13c4-40030-
> 16ba5-2f6d4897-16ba5' in 32000 ms (Method: INVITE)
> Ildvaeg*CLI>
> <--- SIP read from 192.168.27.98:5060 --->
> ACK sip:**1 at 192.168.2.1 SIP/2.0
> From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
> 16ba5-550488a5-16ba5
> To: <sip:**1 at 192.168.2.1>;tag=as3903248a
> Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
> CSeq: 2 ACK
> Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
> 58c7d53-6bede6e9
> Max-Forwards: 70
> Contact: <sip:018 at 192.168.27.98:5060>
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
>
>
>
>
> > Check your features.conf file and make sure that combination or a
> > similar combination (*1 or ** ) for instance isn't defined in there for
> > some reason.
> >
> >
> > Matt Watson wrote:
> > > Might want to do a "sip set debug peer <peer id>"
> > >
> > > You should then be able to see the sip packet dumps that are going
> between the phone and *. Might give you some clues.
> > >
> > > --
> > > Matt
>
> -----------------------------------------------------------
> Henrik ¥stergaard Madsen Phone: +45 44 48 44 92
> PhD, M.Sc. Cell: +45 30 94 02 88
> Mosegard Park 42 email: Henrik at Ostergaard.net
> DK-3500 Værl¢se WWW homepage:
> Denmark http://www.Ostergaard.net/Henrik
>
>
>
>
>
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Best Regards
Rizwan Hisham
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