Re: [asterisk-users] One server, multiple companies
Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a
gotoif to get to correct greeting (correct company).
My question is... lets assume all three companies have extension numbers
being 2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I use context
correctly so that the user at 4002 cannot get through to the user at company
A whose extension is 2000 as currently, I can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my architecture
correct? Should I be running different processes for each company? Can
context resolve what I need?
Please advise.
thanks,
Otto
First off, *nuke* the default context in sip.conf, extensions.conf, and
voicemail.conf ... it will just get you into trouble!
I do something like in my extensions.conf file:
[incoming]
exten = 208229,1,Goto(s,1,incoming-acme)
exten = 208229,1,Goto(s,1,incoming-fido)
exten = 208229,1,Goto(s,1,incoming-big-jims)
...
[incoming-acme]
exten = s,1,Answer()
exten = s,n,Wait(0.75)
exten = s,n(greeting),Playback(brief-directory-acme)
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
; these are the extensions that are exposed both to internal callers as
; well as to incoming calls... be careful what you put here.
include = extens-acme
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[internal-acme]
exten = s,1,Answer()
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
include = outbound-toll
include = outbound-local
include = extens-acme
; for our SIP phones, we can program a non-numeric extension
exten = voicemail,1,VoicemailMain([EMAIL PROTECTED])
exten = voicemail,n,Hangup()
; and for DTMF coming through an ATA...
exten = 777,1,Goto(voicemail)
[extens-acme]
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
...
[outbound-local]
exten = _NXX,1,Dial(${TRUNK}/${AREA}${EXTEN},,r)
exten = _NXX,n,Congestion()
exten = _NXX,n,Hangup()
[outbound-toll]
exten = _NX,1,Dial(${TRUNK}/${EXTEN},,r)
exten = _NX,n,Congestion()
exten = _NX,n,Hangup()
exten = _011.,1,Dial(${TRUNK}/${EXTEN:3},,r)
exten = _011.,n,Congestion()
exten = _011.,n,Hangup()
Note: we had to modify the stdexten macro to be:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
[asterisk-users] One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a gotoif
to get to correct greeting (correct company).
My question is... lets assume all three companies have extension numbers being
2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I use context
correctly so that the user at 4002 cannot get through to the user at company A
whose extension is 2000 as currently, I can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my architecture
correct? Should I be running different processes for each company? Can context
resolve what I need?
Please advise.
thanks,
Otto
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[Asterisk-Users] [perhpas OT] asterisk holding rtp ports open with natted spa-3000
0.0.0.0:10073 0.0.0.0:*udp 0 0 my.public.ip.address:10078 0.0.0.0:* udp0 0 0.0.0.0:10079 0.0.0.0:* [3] Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-3c9c7b77;received=my.wan.ip.address;rport=5060 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=as4cef267a Call-ID: [EMAIL PROTECTED] CSeq: 5600 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 10 Contact: sip:[EMAIL PROTECTED]:5060;expires=10 Date: Fri, 19 Nov 2004 03:16:43 GMT Content-Length: 0 to my.wan.ip.address:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms mercury*CLI Sip read: REGISTER sip:my.asterisk.fqdm SIP/2.0 Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-46a3be23 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: Eric C. Snowdeal III sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5601 REGISTER Max-Forwards: 70 Authorization: Digest username=2000,realm=asterisk,nonce=175e9928,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=81f28467735f8c8676486d56335a2d05 Contact: Eric C. Snowdeal III sip:[EMAIL PROTECTED]:5060;expires=10 User-Agent: Sipura/SPA3000-2.0.11(GWg) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 13 headers, 0 lines Using latest request as basis request Sending to my.wan.ip.address : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-46a3be23;received=my.wan.ip.address;rport=5060 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=as4cef267a Call-ID: [EMAIL PROTECTED] CSeq: 5601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 10 Contact: sip:[EMAIL PROTECTED];expires=10 Content-Length: 0 to my.wan.ip.address:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-46a3be23;received=my.wan.ip.address;rport=5060 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=as4cef267a Call-ID: [EMAIL PROTECTED] CSeq: 5601 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 10 Contact: sip:[EMAIL PROTECTED]:5060;expires=10 Date: Fri, 19 Nov 2004 03:16:52 GMT Content-Length: 0 to my.wan.ip.address:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms mercury*CLI Sip read: NOTIFY sip:my.asterisk.fqdm SIP/2.0 Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-a0bf7d74 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: sip:my.asterisk.fqdm Call-ID: [EMAIL PROTECTED] CSeq: 2999 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA3000-2.0.11(GWg) Content-Length: 0 10 headers, 0 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip.address:5060;branch=z9hG4bK-a0bf7d74 From: Eric C. Snowdeal III sip:[EMAIL PROTECTED];tag=2fc5ef1bb4791707o0 To: sip:my.asterisk.fqdm;tag=as61650e74 Call-ID: [EMAIL PROTECTED] CSeq: 2999 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to my.wan.ip.address:5060 Destroying call '[EMAIL PROTECTED]' [4] http://voxilla.com/forum-viewtopic-t-1290.html [5] Nov 18 19:15:49 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:16:14 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:16:39 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:17:19 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:18:13 NOTICE[18956]: chan_sip.c:6606 handle_response: Peer '2000' is now REACHABLE! Nov 18 19:19:15 NOTICE[18956]: chan_sip.c:6612 handle_response: Peer '2000' is now TOO LAGGED! Nov 18 19:19:49 NOTICE[18956]: chan_sip.c:6606 handle_response: Peer '2000' is now REACHABLE! Nov 18 19:21:51 NOTICE[18956]: chan_sip.c:6612 handle_response: Peer '2000' is now TOO LAGGED! Nov 18 19:22:16 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:22:52 NOTICE[18956]: chan_sip.c:7911 sip_poke_noanswer: Peer '2000' is now UNREACHABLE! Nov 18 19:23:02 NOTICE[18956]: chan_sip.c:6606 handle_response: Peer '2000' is now REACHABLE! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
Eric C. Snowdeal III wrote: after registering the phones correctly and receiving a 200 o.k. message i can connect to other registered softphones and pstn endpoints [ via an voicepulse account ], but after making the initial connection, i can't hear any sound and i get disconnected after getting the following error message: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 41270 prompted by a recent email to the group [1] about setting the bindaddr, i took a closer look at the sip messages being sent back and forth and noticed that the contact header was incorrectly set to 127.0.0.1 in the 200 o.k. message [2]. once i set the bindaddr to the * machine's public ip address everything worked fine and and contact header i.p. address was set correctly. what's odd, at least to me, is that unlike the recent email about a similar issue [1], my * box is on a non-natted, public ip address so i would have thought that keeping the default bindaddr (0.0.0.0) would have worked, but obviously it didn't. not sure how to interpret the dirth of responses, perhaps this was frighteningly obvious to everyone else. [1] http://lists.digium.com/pipermail/asterisk-users/2004-June/051375.html [2] RECEIVE TIME: 7548279 RECEIVE my.public.asterisk.ip:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip:5060;rport;branch=z9hG4bKFE76D502C2BF11D8A741000D93C22AF4;received=my.wan.ip;rport=1029 From: snowdeal sip:[EMAIL PROTECTED];tag=1666554831 To: snowdeal sip:[EMAIL PROTECTED];tag=as7f7ed33f Call-ID: [EMAIL PROTECTED] CSeq: 43970 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 1800 Contact: sip:[EMAIL PROTECTED];expires=1800 Date: Sun, 20 Jun 2004 13:44:34 GMT Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
Holger Schurig wrote: i'm new to asterisk and am having trouble placing outbound calls. i Bug Grandstream so that they finally fix their buggy software. The GS phone sends occassional SIP packets to port 0, not to port 5060, as tcpdump or (better) ethereal will show you. There's a page on this at voip-info.org. thanks for the heads-up about grandstream, but as i stated in the original message, i'm using xten lite softphones. hopefully this is the approproriate forum for this question; i believe this is not an xten configuration issue because i can connect to a ser/rtproxy/nathelper server without problems and i can connect directly to a voicepulse account, which leads me to believe that this is an * configuration problem on my part. less likely, i suppose, is the chance that * isn't as robust in handling nat than ser or whatever voicepulse is running. given the configuration files that i posted in the original message, are there any changes that i should make? certainly the asterisk faq makes the solution seems straighforward [1]: Most likely you have a SIP client behind NAT that is trying to communicate with Asterisk without having the nat=yes setting in place in sip.conf. Another cause for this could be related to a user device that has an sip entry but has been physically removed (switched off or LAN-disconnected). but as my original message showed, i do have nat=yes in my sip.conf and i don't believe the latter scenario is true. any help is greatly appreciated. [1] http://www.voip-info.org/wiki-Asterisk+FAQ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
i'm new to asterisk and am having trouble placing outbound calls. i know this topic has been discussed ad nauseum in the past [1] , but i can't seem to find a workaround and i'm wondering if my newbie-ness is getting the best of me. after registering the phones correctly and receiving a 200 o.k. message i can connect to other registered softphones and pstn endpoints [ via an voicepulse account ], but after making the initial connection, i can't hear any sound and i get disconnected after getting the following error message: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 41270 i've compiled the stock asterisk tarball on a redhat 7.3 box with a public ip address. the clients are xten lite softphone's running on ibooks with os 10.3.4. the clients are natted behind a linksys wrt54g wireless router running the sveasoft [2] firmware. i'm perplexed, because i can get things to work fine if i use ser/rtpproxy instead of asterisk. i can also connect directly to my voicepulse connect account with the xten softphone and things work great. so i think i have the xten client configured properly and i know that the sveasoft firmware isn't throwing a monkey wrench into the picture. i suppose i could configure ser to front asterisk since it appears to deal with the nat, but i'm wondering if i'm missing something basic. my channel config files look like the following: sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=supersecret ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no; Typically set to NO if behind NAT qualify=500 [2001] type=friend ; This device takes and makes calls username=2001 ; Username on device secret=supersecret2 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no; Typically set to NO if behind NAT qualify=500 iax.conf [general] port=5036 bandwidth=low disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference jitterbuffer=no [voicepulse] context = voicepulse-in secret=topsecrect auth=md5 type=friend host=gw5.voicepulse.com [1] http://www.google.com/search?q=retrans_pkt:+Maximum+retries+exceeded+on+call++site:http://lists.digium.comhl=enlr=ie=UTF-8start=10sa=N [2] http://www.sveasoft.com/modules/phpBB2/index.php ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
