Re: [asterisk-users] adding a second extension
And this phone are connected in a local LAN?? Because I see Asterisk receiving a Bad request from 68.156.63.118 If those phones are not in your local LAN, try with a soft phone first. Could be Zoiper or Xlite. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a 400 Bad request back to Asterisk. On Wed, Oct 22, 2008 at 9:10 PM, Stephen Reese [EMAIL PROTECTED] wrote: On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: What kind of phone are you trying to connect to 101??? and from where? Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can contact 102 by dialing 101 but not the other way around, I just get a busy tone. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: And this phone are connected in a local LAN?? Because I see Asterisk receiving a Bad request from 68.156.63.118 If those phones are not in your local LAN, try with a soft phone first. Could be Zoiper or Xlite. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a 400 Bad request back to Asterisk. Both of these phones are on my local lan but the Asterisk server is at a colo facility on the internet outside of the local lan. The local lan does use NAT/PAT. I see an error Warning: 399 Bad Request - 'Malformed/Missing FROM: field'. Is this a problem? Thanks --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 274 Content-Type: application/sdp v=0 o=102 157742 157742 IN IP4 172.16.2.18 s=Cisco 7912 SIP Call c=IN IP4 68.156.63.118 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 - --- (14 headers 12 lines) --- Sending to 68.156.63.118 : 1083 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (NAT) to 68.156.63.118:1082 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net, nonce=7c2e1ba9 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '102' --- SIP read from 64.2.142.116:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 - --- (10 headers 0 lines) --- --- SIP read from 64.2.142.116:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060 From: sip:[EMAIL PROTECTED];tag=as401a34d4 To: sip:[EMAIL PROTECTED];tag=as7a2f92a1 Call-ID: [EMAIL PROTECTED] CSeq: 3064 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec Content-Length: 0 - --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name inbound18.vitelity.net REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 64.2.142.116:5060: REGISTER sip:inbound18.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport From: sip:[EMAIL PROTECTED];tag=as751cb0af To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 3065 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=rsreese, realm=asterisk, algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec, response=b765dbdebba8af18b19707efe651d65d Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from 68.156.63.118:1082 --- ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Content-Length: 0 - --- (9 headers 0 lines) --- ns1*CLI --- SIP read from 68.156.63.118:1082 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2 From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest
Re: [asterisk-users] adding a second extension
I am able to now call the second extension when setup like this so I
believe I'll leave it alone for a while. Basically added the extension
102 to the main incoming line:
exten = 101,1,Dial(SIP/101SIP/102SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,Dial(SIP/102,20)
exten = 102,n,Hangup
exten = 102,n,Voicemail([EMAIL PROTECTED])
Both extensions can call each other and both extensions ring when the
main line is called... Strange but whatever.
On Thu, Oct 23, 2008 at 1:47 PM, Stephen Reese [EMAIL PROTECTED] wrote:
On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
And this phone are connected in a local LAN??
Because I see Asterisk receiving a Bad request from 68.156.63.118
If those phones are not in your local LAN, try with a soft phone first.
Could be Zoiper or Xlite.
Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
sending a 400 Bad request back to Asterisk.
Both of these phones are on my local lan but the Asterisk server is at
a colo facility on the internet outside of the local lan. The local
lan does use NAT/PAT. I see an error Warning: 399 Bad Request -
'Malformed/Missing FROM: field'. Is this a problem?
Thanks
---
ns1*CLI
--- SIP read from 68.156.63.118:1082 ---
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157742 157742 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (14 headers 12 lines) ---
Sending to 68.156.63.118 : 1083 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
--- Reliably Transmitting (NAT) to 68.156.63.118:1082 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
From: sip:[EMAIL PROTECTED];user=phone;tag=2678814914
To: sip:[EMAIL PROTECTED];user=phone;tag=as355e0f84
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=ns1.neocipher.net,
nonce=7c2e1ba9
Content-Length: 0
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000
ms (Method: INVITE)
Found user '102'
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
-
--- (10 headers 0 lines) ---
--- SIP read from 64.2.142.116:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: sip:[EMAIL PROTECTED];tag=as401a34d4
To: sip:[EMAIL PROTECTED];tag=as7a2f92a1
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=575628ec
Content-Length: 0
-
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name
inbound18.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.116:5060:
REGISTER sip:inbound18.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
From: sip:[EMAIL PROTECTED];tag=as751cb0af
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=rsreese, realm=asterisk,
algorithm=MD5, uri=sip:inbound18.vitelity.net, nonce=575628ec,
response=b765dbdebba8af18b19707efe651d65d
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
---
--- SIP read from 68.156.63.118:1082 ---
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP
Re: [asterisk-users] adding a second extension
I also tried downgrading to version 1.4-current but that didn't help. Any other ideas? I'm at a loss. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
What kind of phone are you trying to connect to 101??? and from where? On Wed, Oct 22, 2008 at 7:07 PM, Stephen Reese [EMAIL PROTECTED] wrote: I also tried downgrading to version 1.4-current but that didn't help. Any other ideas? I'm at a loss. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: What kind of phone are you trying to connect to 101??? and from where? Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can contact 102 by dialing 101 but not the other way around, I just get a busy tone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
I am now using a Cisco phone for the second extension (102). I am able to contact 102 from 101 but not the other way around. The error seems less severe now: == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-0825b118, SIP/101/20) in new stack == Using SIP RTP CoS mark 5 -- Called 101/20 -- SIP/101-08221a78 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-0825b118, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-0825b118' So maybe it's just a config issue now? [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [default] exten = 101,1,Dial(SIP/101/20) exten = 101,n,Hangup exten = 101,n,Voicemail([EMAIL PROTECTED]) exten = 102,1,Dial(SIP/102,20) exten = 102,n,Hangup exten = 102,n,Voicemail([EMAIL PROTECTED]) exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) include = inbound include = outgoing [inbound] exten = 9045622082,1,Goto(default,101,1) [outgoing] exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082) exten = _1NXXNXX,n,Set(CALLERID(name)=Stephen Reese) exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]) exten = _NXX,1,Set(CALLERID(num)=9045622082) exten = _NXX,n,Set(CALLERID(name)=Stephen Reese) exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED]) exten = _NXXNXX,1,Set(CALLERID(num)=9045622082) exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese) exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]) exten = _011.,1,Set(CALLERID(num)=9045622082) exten = _011.,n,Set(CALLERID(name)=Stephen Reese) exten = _011.,n,Dial(SIP/[EMAIL PROTECTED]) exten = _911,1,Set(CALLERID(num)=9045622082) exten = _911,n,Set(CALLERID(name)=Stephen Reese) exten = _911,n,Dial(SIP/[EMAIL PROTECTED]) On Mon, Oct 20, 2008 at 11:06 AM, Stephen Reese [EMAIL PROTECTED] wrote: On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez [EMAIL PROTECTED] wrote: I do not think NAT is the problem, NAT normally gives you problems like one way audio or no registration. Try calling the SIP/102 on other extension: ;TEST exten = 1002,1,Dial(SIP,102|20) exten = 1002,n,Hangup() instead of: exten = 102,1,Dial... But this is a very strange error... Check if there is no other definition of default having 102 on it because Asterisk is going to merge the extensions. I get the following when trying to dial 1002 from 101. I've attached my extensions.conf file in-case there is something else that is conflicting as you mentioned. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-082aca90, SIP/102/20) in new stack == Using SIP RTP CoS mark 5 -- Called 102/20 [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-082aca90, ) in new stack == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Tue, Oct 21, 2008 at 9:56 AM, Juan Rodríguez [EMAIL PROTECTED] wrote: Try changing: exten = 101,1,Dial(SIP/101/20) to exten = 101,1,Dial(SIP/101|20) or exten = 101,1,Dial(SIP/101,20) because exten = 101,1,Dial(SIP/101/20) means you are trying to contact ext. 20 on through a trunk called 101. Oh, typo, but that still didn't cure it Successful call from from 101 to 102 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08220318, SIP/102,20) in new stack == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-08221a78 is ringing -- SIP/102-08221a78 answered SIP/101-08220318 -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78 == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318' Failed call from 102 to 101 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08221a78, SIP/101,20) in new stack == Using SIP RTP CoS mark 5 -- Called 101 -- Got SIP response 400 Bad Request back from 68.156.63.118 -- SIP/101-0821e110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08221a78, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
I also tried downgrading to version 1.4-current but that didn't help. Oh, typo, but that still didn't cure it Successful call from from 101 to 102 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08220318, SIP/102,20) in new stack == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-08221a78 is ringing -- SIP/102-08221a78 answered SIP/101-08220318 -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78 == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318' Failed call from 102 to 101 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08221a78, SIP/101,20) in new stack == Using SIP RTP CoS mark 5 -- Called 101 -- Got SIP response 400 Bad Request back from 68.156.63.118 -- SIP/101-0821e110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08221a78, ) in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Mon, Oct 20, 2008 at 12:25 AM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
I'm surprised nobody else noticed this.
I looked through my extensions.conf and sip.conf which are posted in
this thread I believe and didn't turn up anything significant? Would
NAT pose a problem for more then one phone behind a NAT router?
___
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Re: [asterisk-users] adding a second extension
On Mon, Oct 20, 2008 at 12:23 AM, Juan Rodríguez [EMAIL PROTECTED] wrote: The second call its OK, so the problem it is just with the Dial(SIP/102), so try: originate SIP/102 application Dial SIP/102 and originate SIP/101 application Dial SIP/102 and originate SIP/102 application Dial SIP/101 ns1*CLI originate SIP/102 application Dial SIP/102 ns1*CLI == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/102-0824a330 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-0824a330 requested special control 16, passing it to SIP/102-082256c0 -- Started music on hold, class 'default', on SIP/102-082256c0 -- SIP/102-082256c0 answered SIP/102-0824a330 -- Packet2Packet bridging SIP/102-0824a330 and SIP/102-082256c0 -- Stopped music on hold on SIP/102-082256c0 ns1*CLI originate SIP/101 application Dial SIP/102 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/101-08249e28 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-082256c0 answered SIP/101-08249e28 -- Packet2Packet bridging SIP/101-08249e28 and SIP/102-082256c0 ns1*CLI originate SIP/102 application Dial SIP/101 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/101) on SIP/102-08254038 == Using SIP RTP CoS mark 5 -- Called 101 -- SIP/101-08252a40 is ringing -- SIP/101-08252a40 answered SIP/102-08254038 -- Packet2Packet bridging SIP/102-08254038 and SIP/101-08252a40 So I the two extensions are able to call each other with the later two sets of commands so there is hope :-). Would my NAT have anything to do with it since I'm specifying the proxy host that is outside of my firewall? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
I do not think NAT is the problem, NAT normally gives you problems like one
way audio or no registration.
Try calling the SIP/102 on other extension:
;TEST
exten = 1002,1,Dial(SIP,102|20)
exten = 1002,n,Hangup()
instead of:
exten = 102,1,Dial...
But this is a very strange error... Check if there is no other definition of
default having 102 on it because Asterisk is going to merge the extensions.
On Mon, Oct 20, 2008 at 10:09 AM, Stephen Reese [EMAIL PROTECTED] wrote:
On Mon, Oct 20, 2008 at 12:25 AM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
I'm surprised nobody else noticed this.
I looked through my extensions.conf and sip.conf which are posted in
this thread I believe and didn't turn up anything significant? Would
NAT pose a problem for more then one phone behind a NAT router?
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Re: [asterisk-users] adding a second extension
On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez [EMAIL PROTECTED] wrote: I do not think NAT is the problem, NAT normally gives you problems like one way audio or no registration. Try calling the SIP/102 on other extension: ;TEST exten = 1002,1,Dial(SIP,102|20) exten = 1002,n,Hangup() instead of: exten = 102,1,Dial... But this is a very strange error... Check if there is no other definition of default having 102 on it because Asterisk is going to merge the extensions. I get the following when trying to dial 1002 from 101. I've attached my extensions.conf file in-case there is something else that is conflicting as you mentioned. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-082aca90, SIP/102/20) in new stack == Using SIP RTP CoS mark 5 -- Called 102/20 [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-082aca90, ) in new stack == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90' extensions.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
;include = demo
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,Dial(SIP/102,20)
exten = 102,n,Hangup
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,101,1)
[outgoing]
; The following gives an Unknown Caller ID
;exten = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXX,1,Set(CALLERID(num)=9045622082)
exten = _NXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
This is a call from extension 101 to 102 that fails with a busy signal.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08266f60,
'SIP/102',20) in new stack
[Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08266f60, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60'
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Re: [asterisk-users] adding a second extension
What is vitel-outbound?? an IP address??
And what version of Asterisk is this?
Regards,
Juan
On Sun, Oct 19, 2008 at 3:30 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
;include = demo
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
exten = 102,1,Dial(SIP/102,20)
exten = 102,n,Hangup
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,101,1)
[outgoing]
; The following gives an Unknown Caller ID
;exten = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXX,1,Set(CALLERID(num)=9045622082)
exten = _NXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
This is a call from extension 101 to 102 that fails with a busy signal.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08266f60,
'SIP/102',20) in new stack
[Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08266f60, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60'
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Re: [asterisk-users] adding a second extension
On Sun, Oct 19, 2008 at 3:55 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: What is vitel-outbound?? an IP address?? And what version of Asterisk is this? Regards, Juan vitel-outbond is the connection to my sip provider Version 1.6 of Asterisk I'm able to make incoming and outgoing calls just fine using 101. It's the extension to extension calling from 101 to 102 and vice-versa that is not working. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
First, I think is better to to have SIP/vitel-outbound/${EXTEN} than
having SIP/[EMAIL PROTECTED]
And try issuing SIP SET DEBUG on the cli to see what happens when making the
call, post back what you see making calls from 101 to 102 and 102 to 101.
Having the sip.conf sould help on getting whats going on.
Here are the relevant parts of the sip.conf
[general]
register = rsreese:[EMAIL PROTECTED]:5060/rsreese
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
;match_auth_username=yes; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
;allowtransfer=no ; Disable all transfers (unless
enabled in peers or users)
; Default is enabled
realm=ns1.neocipher.net ; Realm for digest authentication
; defaults to asterisk. If you set a
system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard
port for unencrypted UDP
; and TCP sessions is 5060)
; bindport is the local UDP port that
Asterisk will listen on
bindaddr=0.0.0.0; IP address to bind UDP listen socket
to (0.0.0.0 binds to all)
; You can specify port here too, like
123.123.123.123:5080
domain=neocipher.net
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=pass
mailbox=101
callerid=\Stephen\ 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=pass
mailbox=102
callerid=\Stephen\ 102
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=test
allow=all
;insecure=very
insecure = invite
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=test
allow=all
canreinvite=no
Here is the sip debug error:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08266f60,
'SIP/102',20) in new stack
[Oct 19 16:21:21] WARNING[26690]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 16:21:21] WARNING[26690]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08266f60, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/102-08266f60'
Scheduling destruction of SIP dialog
'NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.' in 32000 ms (Method:
INVITE)
ns1*CLI
--- Reliably Transmitting (NAT) to 68.156.63.118:56558 ---
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
68.156.63.118:56558;branch=z9hG4bK-d8754z-e0a3d830adb35401-1---d8754z-;received=68.156.63.118;rport=56558
From: sip:[EMAIL PROTECTED];tag=7d39014c
To: 102sip:[EMAIL PROTECTED];tag=as4f32f2a7
Call-ID: NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
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Re: [asterisk-users] adding a second extension
On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: Try reinstalling Asterisk, because in the channel.c this error is returned if the channels TEC (in this case SIP) is not found. Weird!! Let me know if it works. Regards, Juan So the extensions.conf and sip.conf look correct? I tried reinstalling and I still am unable to communicate between the two extensions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
Stephen: Your configuration files looks fine. Try from the CLI issuing originate SIP/101 extension [EMAIL PROTECTED], having the 101 online, then do that with originate SIP/102 extension [EMAIL PROTECTED]. See what happens. If you got a CVS commit, commit again or try installing a release. http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for download) Regards, Juan On Sun, Oct 19, 2008 at 10:36 PM, Stephen Reese [EMAIL PROTECTED] wrote: On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: Try reinstalling Asterisk, because in the channel.c this error is returned if the channels TEC (in this case SIP) is not found. Weird!! Let me know if it works. Regards, Juan So the extensions.conf and sip.conf look correct? I tried reinstalling and I still am unable to communicate between the two extensions. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: Stephen: Your configuration files looks fine. Try from the CLI issuing originate SIP/101 extension [EMAIL PROTECTED], having the 101 online, then do that with originate SIP/102 extension [EMAIL PROTECTED]. See what happens. If you got a CVS commit, commit again or try installing a release. http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for download) Regards, Juan I grabbed the latest tarball and installed it. The extension rings through to 101 and then when I answer it tries to ring through to 102 but seems to fail. ns1*CLI originate SIP/101 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08245390, 'SIP/102',20) in new stack [Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08245390, ) in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390' The extension rings through to 102 and when I answer the line it begins to ring line 101. ns1*CLI originate SIP/102 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08249e28, SIP/101SIP/[EMAIL PROTECTED],30) in new stack == Using SIP RTP CoS mark 5 -- Called 101 == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- SIP/101-08244e88 is ringing -- SIP/vitel-outbound-0825d1e0 is making progress passing it to SIP/102-08249e28 -- SIP/vitel-outbound-0825d1e0 is ringing -- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28 -- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0 == Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28' I'm at a loss. Thanks for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
The second call its OK, so the problem it is just with the Dial(SIP/102), so try: originate SIP/102 application Dial SIP/102 and originate SIP/101 application Dial SIP/102 and originate SIP/102 application Dial SIP/101 On Sun, Oct 19, 2008 at 11:46 PM, Stephen Reese [EMAIL PROTECTED] wrote: On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: Stephen: Your configuration files looks fine. Try from the CLI issuing originate SIP/101 extension [EMAIL PROTECTED], having the 101 online, then do that with originate SIP/102 extension [EMAIL PROTECTED]. See what happens. If you got a CVS commit, commit again or try installing a release. http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz(for download) Regards, Juan I grabbed the latest tarball and installed it. The extension rings through to 101 and then when I answer it tries to ring through to 102 but seems to fail. ns1*CLI originate SIP/101 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08245390, 'SIP/102',20) in new stack [Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08245390, ) in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390' The extension rings through to 102 and when I answer the line it begins to ring line 101. ns1*CLI originate SIP/102 extension [EMAIL PROTECTED] == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08249e28, SIP/101SIP/[EMAIL PROTECTED],30) in new stack == Using SIP RTP CoS mark 5 -- Called 101 == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- SIP/101-08244e88 is ringing -- SIP/vitel-outbound-0825d1e0 is making progress passing it to SIP/102-08249e28 -- SIP/vitel-outbound-0825d1e0 is ringing -- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28 -- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0 == Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28' I'm at a loss. Thanks for your help. -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding a second extension
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
I'm surprised nobody else noticed this.
Stephen Reese wrote:
On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Stephen:
Your configuration files looks fine. Try from the CLI issuing originate
SIP/101 extension [EMAIL PROTECTED], having the 101 online, then do that
with
originate SIP/102 extension [EMAIL PROTECTED]. See what happens.
If you got a CVS commit, commit again or try installing a release.
http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for
download)
Regards,
Juan
I grabbed the latest tarball and installed it.
The extension rings through to 101 and then when I answer it tries to
ring through to 102 but seems to fail.
ns1*CLI originate SIP/101 extension [EMAIL PROTECTED]
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-08245390,
'SIP/102',20) in new stack
[Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/101-08245390, ) in new
stack
== Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390'
The extension rings through to 102 and when I answer the line it
begins to ring line 101.
ns1*CLI originate SIP/102 extension [EMAIL PROTECTED]
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-08249e28,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-08244e88 is ringing
-- SIP/vitel-outbound-0825d1e0 is making progress passing it to
SIP/102-08249e28
-- SIP/vitel-outbound-0825d1e0 is ringing
-- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28
-- Packet2Packet bridging SIP/102-08249e28 and SIP/vitel-outbound-0825d1e0
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28'
I'm at a loss. Thanks for your help.
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