[Asterisk-Users] how to tell what ${DIALSTATUS} is being set
i often have nufone problems, e.g.
-- Executing Dial(SIP/konaa0p-4b88, IAX2/[EMAIL PROTECTED]/14086661234)
in new stack
-- Called [EMAIL PROTECTED]/14086661234
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/NuFone/5'
sound of surf (on a boogie board kind of day) for a fairly long while
== No one is available to answer at this time
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
== Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
-- Executing Hangup(SIP/konaa0p-4b88, ) in new stack
== Spawn extension (dial-gateways, h, 1) exited non-zero on 'SIP/konaa0p-4b88'
i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them. the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing. with BUSY vs CONGESTION this is even more of an
issue.
is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?
randy
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[Asterisk-Users] Re: Encrypted VOIP?
Just run point to point encryption over a vpn. Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious. classic problem. how do you know, in a way that the application and user can see it, that the data are on a crypted channel? this is a problem in general with all the rfcs which say for privacy, run it over ipsec. there is no signaling from the transport to the app. randy --- Q: Because it reverses the logical flow of conversation. A: Why is top posting frowned upon? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco phones config over internet
man tftpd, particularly the -c option randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sporadic beeps spa3k-*
freebsd quite current ports tree 1.01 asterisk spa3k at 2.0.11(GWg) for calls in from the pstn side of an spa3k to asterisk, i get sporadic short beeps. they are not related to sip re-reg time, which is all that has occurred to me so far. calls in from the fxs side of the spa3k and out through nufone do not exhibit the beeps. calls from the fxs side of the spa3k out the fxo side do have the beeps. nothing googling the wiki or the net. clues solicited randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 3000 inbound FXO problem
you have two 'friend' entries in your sip.conf. it uses the second, which is not what you want. one should be peer and the other user. though a number of versions of asterisk don't actually work with peer/user, a major pita. so try reversing the order of the two entries if you have problems with peer/user. and don't post a bug report about this. you will get screamed at and insulted (seems the mentality of the asterisk community), and told you should have posted your question to this list despite your already having done so. this kind of response/support is why we went with a commercial solution for production; though i keep * for home use. randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FREE BSD
anynody knows if I Can install and run Asterisk under Free BSD? /usr/ports/net/asterisk randy ___ 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] Re: Cisco Router FXO / Skinny
It would be nice to see a WIKI page on ISDN/PSTN-CISCO-Asterisk. drool! i am having voice quality issues when sipping out over a 7650 fxo to pstn. i sounds a bit like too much silence suppression. randy ___ 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] Re: Asterisk on FreeBSD
I'm very interested if somebody using asterisk on FreeBSD and not Linux without problem ? many of us are using * on 5.3-stable and 6.0-current. without a problem would be a bit pollyanna-like. randy ___ 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] long list of prefixes
if a phone number starts with one of 50+ prefixes, i want to send the sip call to gateway X. if it is in any other prefix, i want to send it to gate Y. i am not excited about a long list of extens, but will do it if i have to. i suspect there is a database hack, but i lose all database contents if i reinstall the port (this may be a feature of the freebsd port), and i have not figured out a script that will let me load it. surely there is a well-known and reasonable way out of this corner. but i can not seem to find the right wiki incantation. thanks for clue. randy ___ 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] Re: long list of prefixes
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc
too big a hammer. i finally did the agi hack. for the archive
[dial-hawi]
exten = s,1,NoOp(dial-hawi)
exten = _.,1,SetVar(PREFIX=)
exten = _.,2,AGI(agi-prefix|${EXTEN:4:3})
exten = _.,3,NoOp(agi-prefix returns ${PREFIX})
exten = _.,4,Dial(SIP/${PREFIX}${EXTEN:[EMAIL PROTECTED],60,Ttr)
exten = h,1,Hangup()
exten = i,1,GoTo(s,1)
exten = t,1,GoTo(s,1)
with the script being a brutal
#!/usr/local/bin/bash
if ! grep $1 /usr/local/etc/hawi-prefixes /dev/null; then
echo SET VARIABLE PREFIX 1808
fi
randy
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[Asterisk-Users] Re: Advantage of IAX2 to SIP?
Some - few - providers are using IAX2 as a protocol. Most are using SIP. I know that there are advantages of IAX2 regarding multiple connections. But beside this I'm asking myself (and you all) why I should prefer IAX2 when my SIP connection is working. some discussion of this a few months back produced the following excellent precis From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: Re: iax or sip Date: Mon, 5 Jul 2004 18:59:52 -0500 (CDT) Okay, setting aside conspiracy theories, trolling, flaming, etc, let me summarize some differences between SIP and IAX, and it might help you make a decision about what is best for you. 1) IAX is more efficient on the wire than RTP for *any* number of calls, *any* codec. The benefit is anywhere from 2.4k for a single call to approximately trippling the number of calls per megabit for G.729 when measured to the MAC level when running trunk mode. 2) IAX is information-element encoded rather than ASCII encoded. This makes implementations substantially simpler and more robust to buffer overrun attacks since absolutely no text parsing or interpretation is required. The IAXy runs its entire IP stack, IAX stack, TDM interface, echo canceller, and callerid generation in 4k of heap and stack and 64k of flash. Clearly this demonstrates the implementation efficiency of its design. The size of IAX signalling packets is phenomenally smaller than those of SIP, but that is generally not a concern except with large numbers of clients frequently registering. Generally speaking, IAX2 is more efficient in its encoding, decoding and verifying information, and it would be extremely difficult for an author of an IAX implementation to somehow be incompatible with another implementation since so little is left to interpretation. 3) IAX has a very clear layer2 and layer3 separation, meaning that both signalling and audio have defined states, are robustly transmitted in a consistant fashion, and that when one end of the call abruptly disappears, the call WILL terminate in a timely fashion, even if no more signalling and/or audio is received. SIP does not have such a mechanism, and its reliability from a signalling perspective is obviously very poor and clumsy requiring additional standards beyond the core RF3261. 4) IAX's unified signalling and audio paths permit it to transparently navigate NAT's and provide a firewal administrator only a *single* port to have to open to permit its use. It requires an IAX client to know absolutely nothing about the network that it is on to operate. More clearly stated, there is *never* a situation that can be created with a firewall in which IAX can complete a call and not be able to pass audio (except of course if there was insufficient bandwidth). 5) IAX's authenticated transfer system allows you to transfer audio and call control off a server-in-the-middle in a robust fashion such that if the two endpoints cannot see one another for any reason, the call continues through the central server. 6) IAX clearly separates Caller*ID from the authentication mechanism of the user. SIP does not have a clear method to do this unless Remote-Party-ID is used. 7) SIP is an IETF standard. While there is some fledgling documentation courtesy Frank Miller, IAX is not a published standard at this time. 8) IAX allows an endpoint to check the validity of a phone number to know whether the number is complete, may be complete, or is complete but could be longer. There is no way to completely support this in SIP. 9) IAX always sends DTMF out of band so there is never any confusion about what method is used. 10) IAX support transmission of language and context, which are useful in an Asterisk environment. That's pretty much all that comes to mind at the moment. Mark ___ 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] Re: getting callerid from spa3k to asterisk
if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. aii! randy ___ 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] getting callerid from spa3k to asterisk
ok, with a good pointer from Chris Stenton [EMAIL PROTECTED], i found the problem. if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. so, until the problem is diagnosed, how do i work around it. as the spa3k is registered, i tried to remove the spa3k-out context entirely. callerid now works. yes! but ... if i try to place an outbound call using the spa3k-in context, the call is sent to the spa3k, but it just gives me the pstn's dialtone, and does not dial the number. my spa3k config is in http://rip.psg.com/~randy/spa3k.html. so how do i place a call out the spa3k pstn without a separate outbound context? randy --- Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1 From: CallerName sip:[EMAIL PROTECTED];tag=25aee11517d597a1o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: CallerName sip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: biwa 0431 sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 8805171 8805171 IN IP4 198.180.150.195 s=- c=IN IP4 198.180.150.195 t=0 0 m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 19 lines Using latest request as basis request Sending to 198.180.150.195 : 5061 (non-NAT) Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 198.180.150.195:16396 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'spa3k-out' ___ 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] Re: NoOp
What is the purpose of NoOp (no operation) if it does nothing?
among other things, it logs, so you can see a context being
entered. e.g.
[ext-foo]
exten = _X.,1,NoOp(ext-foo cid=${CALLERIDNUM})
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[Asterisk-Users] Re: getting callerid from spa3k to asterisk
You could maybe look at the autocreatepeer option for sip.conf that level of vulnerability would not seem to be a good approach to solving some sort of sip/config problem :-) the problem is in the sip handshake between the spa3k and *. i have been hoping a sip geek would have a chance to look at it. randy ___ 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] getting callerid from spa3k to asterisk
i am still dying on this one, and my critical user, my fiance'e,
is giving me hell over it on my home test environment; even my
daytime job, for which i am prototyping, is more patient. :-)
i can not get caller-id from a call coming in to the spa3k pstn
to asterisk. fwiw, this used to work with older * and spa3k
versions, but of course it could be something i did to configs.
essentially, if i tell the spa3k to pass callerid to *, the sip
session gets rejected by *. since no one seemed to like to see
ethereal output, i have posted * sip debug form of the sessions.
does anyone have their spa3k and * config working that i could
look at? or, if you can shoot the bug, i'll pay you US$100 by
paypal or whatever.
the spa3k configiuration
http://rip.psg.com/~randy/spa3k.html
sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO
call accepted ok, but no callerid received by asterisk
http://rip.psg.com/~randy/debug-0.txt
sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
call rejected by asterisk
http://rip.psg.com/~randy/debug-1.txt
sip.conf entry
[spa3k-in]
type=friend ; user fails to register
host=dynamic
port=5061
auth=md5
secret=dontbesilly
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=ext-in42
extensions.conf for the incoming
[ext-in42]
exten = _X.,1,NoOp(ext-in42 cid=${CALLERIDNUM})
exten = _X.,2,SetVar(areacode=206)
exten = _X.,3,SetVar(mailbox=1)
exten = _X.,4,GoTo(ext-common,s,1)
[ext-common]
exten = s,1,NoOp(ext-common cid=${CALLERIDNUM})
exten = s,2,Background(zz-who-common)
exten = i,1,Hangup()
exten = t,1,GoTo(ext-common,s,1)
include = speeddials
include = extensions
include = conferences
include = applications
randy
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[Asterisk-Users] Re: How far is IAX to be a Standard
what does the RFC's guys and the Pseudo-Cisco IETF think about this Protocol? the internet vendor task force has a massive amount invested in sip. so there will be a lot of 'guidance' to have it published as an informational rfc. if iax catches on in the market, then they'll have to play. otherwise, expect to have a hard time getting iax on the ivtf standards track. randy ___ 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] Re: Multi-office topology suggestions
The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? for one phone number, look at the sipura spa-3000 or its clones randy ___ 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] Re: Re: doublehash patch for 1.0.1
yes, it is easier to write dialplans with a single #. yes, it is easier to code for a single # transfer. so, if we want a system that is great to write code for and easy to write single-# dialplans for, we're cool to go. otoh, if we want a system that USERS can use, and acts like professional pbxs, i don't think the current method cuts the cake. and products are all about users, pita though they may be. randy ___ 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] Re: doublehash patch for 1.0.1
Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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] getting cid from spa3k pstn to *
in order to get the cid from the spa3k to *, i need to turn on PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES this produces a sip invite as follows: Frame 1 (1092 bytes on wire, 1092 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7 From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: CID Namesip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 430 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol note that the From: has the cid, as does the Remote-Party-ID:. and the Contact: has the spa3k's id and display name. as the sip.conf entry looks like [spa3k] type=friend host=dynamic port=5061 auth=md5 secret=hidden qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=spa3k-ext the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy Authentication Required, to which the spa3k responds Frame 3 (450 bytes on wire, 450 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7 From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1 To: sip:[EMAIL PROTECTED];tag=as2741cf03 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 0 and it all goes to hell from there. if i set the spa3k config to have PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO Frame 1 (1072 bytes on wire, 1072 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a From: spa3k pstn sip:[EMAIL PROTECTED];tag=8fc58211a0dc60f2o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: spa3k pstn sip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 430 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol the connection completes, but asterisk does not have the pstn caller id. randy ___ 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] doublehash patch for 1.0.1
is there a doublehash patch for 1.0.1? o old one to res_parking.c does not apply as there is no longer res_parking.c o wiki search is useless o google only finds the problems applying old patch to 0.7 thanks randy ___ 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] doublehash patch for 1.0.1
and the patch take19.txt in bug 0002010 does not apply cleanly to the freebsd port of 1.0.1 randy ___ 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] spa3k: cid vs authid
freebsd just upgraded to 1.01, thanks sobomax and team! with the upgrade, on inbound from an spa3k pstn call, i started getting the classic Failed to authenticate user callerid when the authenticating client should have been the spa3k/pstn/userid i can get around this by setting the spa3k PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: to NO but now i no longer capture the incoming pstn caller's callerid. what am i not getting here? randy ___ 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] spa3k: cid vs authid
[ replying to myself ]
freebsd just upgraded to 1.01, thanks sobomax and team!
with the upgrade, on inbound from an spa3k pstn call,
i started getting the classic
Failed to authenticate user callerid
when the authenticating client should have been the
spa3k/pstn/userid
i can get around this by setting the spa3k
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID:
to NO
but now i no longer capture the incoming pstn caller's
callerid.
what am i not getting here?
i think i see a glimmer
in asterisk 0.9.0.2, on an incoming pstn call through the
spa3k, the callerid number became usable as the extension
in extensions.conf. so i could recognize, for example,
my mobile phone, callerid=8081234567, in
exten = _#1/8081234567,1,Macro(vmm,s1)
so i could bypass a bunch of authentication crud (until
someone steals my mobile:-).
now, in 1.0.1, this does not seem to work. could it be
the way i am catching the call?
[ext-xxx]
; this catches the spa3k PSTN / Dial Plans / Dial Plan 1: = (S0:123)
exten = s,1,NoOp()
exten = _.,1,SetVar(foo=42)
exten = _.,2,SetVar(bar=666)
exten = _.,3,GoTo(ext-common,${EXTEN},1)
[ext-common]
exten = s,1,Background(zz-get-rid-of-george)
exten = i,1,Hangup()
exten = t,1,GoTo(ext-common,s,1)
include = speeddials
include = extensions
include = conferences
include = applications
[applications]
; voicemail
exten = _#1/8081234567,1,Macro(vmm,s1)
...
randy
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[Asterisk-Users] Re: Sipura-3000 - silent dial out on FXO port
It does work but the customer perceives this delayed second DTMF feedback as unprofessional and the sipura as a toy. I wonder if there is anything that can be done to keep the channel to the caller silent until after the Sipura has sent the DTMF out on the PSTN line. Upgrade your firmware to the latest release. They solved that problem in the more recent releases (2.0.10 and above, IIRC). or, if they have a sense of humor, tell them the equally unprofessional cisco 1750 does the same randy ___ 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] Re: Re: SPA3000 as a replacement for X100P
Old SPA-3000 firmware versions had issues with bad echo when raising txgains, apparently it has been greatly reduced, if not fixed in the latest firmware. greatly reduced, yep. fixed, nope. but it's to the level that my wife is only handing me a bug report occasionally. randy ___ 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] Re: SPA-3k outbound calls...
Maybe if the host was specified in sip.conf rather than being listed as dynamic this wouldn't be necessary. yep. again, i have [spa3k-pstn-out] type=peer auth=md5 secret=haha username=asterisk fromuser=asterisk host=spa3k.host.name port=5061 dtmfmode=rfc2833 nat=no canreinvite=yes context=ext [spa3k-pstn-in] type=friend host=dynamic port=5061 auth=md5 secret=haha qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=ext ___ 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] Re: SPA3000 as a replacement for X100P
With mine no echo problem, but the sound level is very low... :/ You have to speak higher to be heard... Raise the TxGain setting on the SPA. and get echo randy ___ 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] Re: passing octothorpe
The standard way to get around this is to use the doublehash (or maybe doublepound but unlikely to be doubleoctothorpe) patch which will allow you to press hash twice for transfer or once to send it to the remote end. IIRC you can also specify the timeout for it to wait for the second hash. aha! and the latest is the year old one at http://www.mail-archive.com/[EMAIL PROTECTED]/msg06524/doublehash.patch? randy ___ 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] passing octothorpe
some conferencing systems want you to hit octothorpe (aka pound, hash, etc.). once connected, i would have expected * to be transparent to all dtmf codes. it seems not to be. wiki has not been helpful, it seems to have most references to do with octothorpe in dial plan. so, what do i not understand? randy ___ 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] cannonicalizing phone num in macro
i am in a macro. ${ARG2} is a phone number, which might be
seven, ten, or eleven digits. i wish to canonicalize it to
be a full 11 digit number. if this was a normal exten, i
would
exten = _1XX,1,GoTo(dial-gateway,${EXTEN},1)
exten = _XX,1,GoTo(dial-gateway,1${EXTEN},1)
exten = _XXX,1,GoTo(pstn,1${areacode}${EXTEN},1)
but how to hack it inside a macro?
randy
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[Asterisk-Users] x-lite and pound key
[ wiki on xten/x-lite gets you to a 5mb pdf which tells you how to do a windows install. deep :-( ] anyone know how to make x-lite be # key transparent, i.e. send the key when it is poked? randy ___ 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] Re: Convert Cisco 7960 to sip
you need to go: v2 v2.1 v2.3 v4.0 v6.0 i recently succeeded in 2, 2.3, 6, 7.1, and am happy with 7.1 on a number of units. v7.0 doesnt work properly i have found could you be more specific? randy ___ 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: Re[2]: [Asterisk-Users] Cisco SIP Phone 7960 DTMF Problem
After discussion I try the following without result: DIALTEMPLATE TEMPLATE MATCH=#... Timeout=5 User=Phone / TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE Then I try, without result too: DIALTEMPLATE TEMPLATE MATCH=*# Timeout=5 User=Phone/ TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE Then I try, without result too: DIALTEMPLATE TEMPLATE MATCH=..# Timeout=5 User=Phone/ TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE DIALTEMPLATE TEMPLATE MATCH=..# Timeout=5 User=Phone Rewrite=*8#/ TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE what do you mean without result? what happens? the first one works for me. i think that you will have to reset the phone for a dialplan change. randy ___ 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] Re: SPA-3000 as a regular Asterisk FXO device?
Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. i have been doing this since early spa3k beta. but i don't see why such kinky rituals are needed. could someone send clue? randy ___ 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] Re: Cisco SIP Phone 7960 DTMF Problem
Dunno if you can change a cisco to not use # to 'send' http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx ___ 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] Re: Astricon Conference Call?
if it is of help, i can handle some internet bandwidth; two stm-1s to a quite underloaded * server, which is on gige. but i can not provide local pstn gating. randy ___ 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] Re: Echo in asterisk phones.
Also beware if checking it in debug mode (like asterisk -vc) Took me awhile to notice it was going away when I started asterisk normally ban head on wall! ai! that hurts. but thank you! randy ___ 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] Re: how to configure my cisco 7960?!
after plugging it in it says Configuring IP - I unlocked it and entered the Network Configuration. I can see the edit-buttons but when I trie to press then it says That key is not active here before you hit network configuration hit **# to unlock the config you may want to read http://www.cisco.com/warp/customer/788/voip/handset_to_sip.html randy ___ 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] Re: spa-3000 review?
Set Admin-Advanced-Line1-DialPlan- ([2-9]xx:@gw0|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) [2-9]xx:@gw0 will send 211,311,411...911 to gw0 which is the local pots port. this is not what i expected. i expected something like ([49]11:@gw0|rest of dial plan) randy ___ 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] Re: VoicePulse changes
the message arrived here some hours after calls through them stopped working. not very professional. there should have been considerable, like multiple days, of overlap. think about the customers who are out of reach of configuring their * server but still rely on the service. randy ___ 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] Re: Re: OT: saving/restoring sipura config
Sorry for this OT but I bet someone here knows if there is a way to save a Sipura 2000 current config and restoring it after a reset. hard as this is to believe, there isn't. major bummer, eh? I believe the Sipura SPA-2000 can be provisioned via files on a TFTP server, which would act as a backup should the box die. I haven't set this up but will do when I get a chance. kinda, sorta. see spc tool. but that begs the question. many of us, especially those from the large scale internet provisioning world expect to be able to get, by secure means (e.g. ssh), the config from a device in a processable format, maintain archives, cvs, diff, generate new configs, ... and upload them back to the devices. see, for example, https://www.shrubbery.net/rancid/, for an open source downloader, differ, archiver many of us use. randy ___ 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] Re: ISP: ATT or Sprint
Thank you, good comment AtT by far on IP quality, We once had a Sprint DS3 on a 100 meg man link. After 5 months of inadequate service, Sprint had 9 engineers trying to tell me that it was my problem I never got a faster download than 3megabit versus a 9 meg link that absolutely screamed from att. Never again will I use Sprint for an IP connection. They oversell many times their bandwidth and the service sucks if there is a problem. perhaps. but it lacks some. from actual research and operator measurements, sprintlink is highly over-provisioned on the north american backbone and to customer aggregation routers. customer links can easily be saturated. and some of their peering links are under-provisioned, as are many providers' in these tight times, att included. randy, who works for their competition ___ 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] Re: OT: saving/restoring sipura config
Sorry for this OT but I bet someone here knows if there is a way to save a Sipura 2000 current config and restoring it after a reset. hard as this is to believe, there isn't. major bummer, eh? randy ___ 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] Re: incoming calls on Cisco 7960
[214] disallow=all allow=ulaw type=friend secret= host=dynamic nat=no dtmfmode=rfc2833 canreinvite=no incominglimit=1 mailbox=214 where is the context= to send it to an incoming context? ___ 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] Re: Re: iax or sip
i buy most of what you say except At that point, virtually every business and every household will have such a box. Couple that with some universal directory facility, ie ENUM and you have got a ubiquitous peer-to-peer telephone network where telcos will have no role to play other than providing the data pipes. why enum? forcing humans to deal with telephone numbers is analogous to asking them to use ip addresses instead of domain names (which are bad enough, but that's another story). do you want to send email to [EMAIL PROTECTED] so why not 'dial' [EMAIL PROTECTED] or whatever? is it the 12-key pad? but maybe i'll be 'dialing' using my computer's keyboard. after i had spent a bunch of time working out and pushing the enum hack (shamelessly stolen from tpc.int), allison mankin hit me with a clue-by-four and asked what was the sense in mapping phone numbers into a name space that already worked and was sufficient. i felt pretty stoopid. randy ___ 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] dialing # on a crisco (was: Divert to arbitrary number)
On a related note, how do you get a Cisco 7940 to dial numbers with a hash in them, instead of just using the hash as a dial key. For example, I have *#21# to check diverts, but the phone will just dial * as soon as you type the # after it. DIALTEMPLATE TEMPLATE MATCH=#... Timeout=5 User=Phone / TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE randy ___ 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] iax or sip
i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. what am i missing here? randy ___ 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] Re: iax or sip
iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. You may indeed loathe NATted networks, but in general they're very hard to avoid. Why would you criticize a protocol for dealing with such a thing efficiently--which, quite famously, SIP does not? i did not criticize the protocol. remember, my question started with i am looking at iax to see if it is applicable to my needs. i don't need nats, nat traversal, nat anything. if i did, iax might well be one of the technologies i would consider. but i don't. Do you know of a successful VoIP protocol that is entirely TCP-based? not currently, though folk are working hard on the congestion friendliness issue. if you're interested, i can point you to the relevant part of the ivtf basement. I would want the PBX in the datastream in cases where multiple endpoint connections would pass through multiple IAX boxen why? and yes, i mean the question. i see setup running through the boxen, of course. i just don't see why you would want the payload to traverse what might be a pretty baroque multi- continental path. i may have big pipes, but the bleedin' speed of light seems not to be very impressed. Perhaps in your case your networks are all public-IP, running on DS3s or OC48s. well not ds3s, stm-1 and above. but i ain't a big fan of wasting bytes. i am also not a fan of triangle routing. and maybe we could avoid the ad homina which seems to be too frequent on this list? randy ___ 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] Re: iax or sip
There are many reasons to have an Asterisk box in a stream: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) hmmm. post setup, which clearly needs to go through all servers (or pbxen) in path, i don't see a win here. send more clue. 2. Provide features (access to PSTN, conference capability, music on hold, call parking, agents and queues. the list goes on and on) that's setup not payload 3. Endpoints (User Agents) MAY not be able to send data streams to each other directly (firewalls or nats in the middle) yes indeed. so one, but likely only one, if they're asterisk, pbx needs to intermediate, not a bunch on a path. And depending upon your view of things (your view might be different than the view of the IT/communications administrator of a large company), using IAX in a geographically distributed use scenario might very well be exactly what you want (use over an encrypted vpn link, etc.) yes, it might be. but as you know, i am a big pipe backbone geek, not an admin of a large distributed company. and i am addict of simple (non-complex, not the presence protocol:-). randy ___ 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] Re: dialing # on a crisco (was: Divert to arbitrary number)
Is it possible to have a speed dial on a cisco 7960 which dials the voice mail number and then dials the extention and password so a user can just push a single button to get their voicemail? see Message Button under http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx ___ 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] Re: Randy Bush is a destructive force with a hidden professional agenda
I have no idea who Randy Bush is but I found it funny the first article I found on him was a presentation on why NAT is evil espically for voice. Now he asserts that NAT traversal is not needed. no, i said i don't need it randy ___ 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] Re: Re: iax or sip
Okay, setting aside conspiracy theories, trolling, flaming, etc, let me summarize some differences between SIP and IAX, and it might help you make a decision about what is best for you. brilliant piece. thanks! damned shame i cound not find it on google/wiki so i did not have to make so much confusion on the list. i suspect many will appreciate your posting it. being a big pipe backbone geek, i will spend bytes in exchange for ease of scalable deployment, security, simplicity, etc. but i do keep an eye on byte count, of only out of antique habits. and there definitely are applications, though not mine of the season, where byte count is darn near king, e.g., wireless, poor countries, ... as i was discussing in side email with a fellow user. [ but, as a backbone isp, i make money for delivering bytes, so maybe my black helicopter friend (who exceedingly mistakenly associated me with icann and other things) can think wasting bytes is a way for me to increase income:-). 2) IAX is information-element encoded rather than ASCII encoded. i have always been of two minds on binary vs ascii. but, if efficiency is a major goal, binary does get a lot of weight. 3) IAX has a very clear layer2 and layer3 separation oo! you know how to sell to an fsm and protocol freak. 4) IAX's unified signalling and audio paths permit it to transparently navigate NAT's and provide a firewal administrator only a *single* port to have to open to permit its use. yes, if i was worried about nats i could not control, this would be a win. otoh, though not in my current game, i am one of those who worries about protocols being congestion friendly, especially as one approaches the customer edge. 5) IAX's authenticated transfer system allows you to transfer audio and call control off a server-in-the-middle in a robust fashion such that if the two endpoints cannot see one another for any reason, the call continues through the central server. s/off a server/through a server/ ? 6) IAX clearly separates Caller*ID from the authentication mechanism of the user. a win. but eventually the call gets to the damned edge, where all these kinky devices want fsk, dmtf, or tibetan prayer flags. end users and their bleedin' devices are the bane of the internet:-). 7) SIP is an IETF standard. While there is some fledgling documentation courtesy Frank Miller, IAX is not a published standard at this time. in my current life, i am not all that impressed by something being a recent ivtf (sic) standard. but i have been there and done that. but i threw the plaque in the garbage. 8) IAX allows an endpoint to check the validity of a phone number to know whether the number is complete, may be complete, or is complete but could be longer. There is no way to completely support this in SIP. having had great fun with various dialplans and number sequence stuff on various kit, i can appreciate this. but it seemed as if most of the problems i ran into in this area were more device config design than protocol. 9) IAX always sends DTMF out of band so there is never any confusion about what method is used. cool. i did get caught by this recently. 10) IAX support transmission of language and context, which are useful in an Asterisk environment. not sure what you mean? end-user human language? hmmm. that may end up as useful in my current scenario. That's pretty much all that comes to mind at the moment. not bad. and exceedingly helpful! owe you much sushi if we ever run into eachother. randy ___ 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] Re: iax or sip
1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) hmmm. post setup, which clearly needs to go through all servers (or pbxen) in path, i don't see a win here. send more clue. hide one end from the other. I have a customer and a carrier. I don't want one to know who the other is lest they get together and cut me out of the equation. yikes! despite ad homina on this list, even i am not that sneaky. but i can see folk having legitimate needs such as this in an emerging market in desperate times. My comment above not withstanding, might I be correct that your purpose is more along the lines of a personal comm system? while i have that going on the side for fun, see appended, the use for which i am scratching my head is big pipe global backbone stuff. i am in the commercial world in my daytime job. i sold my soul long ago; had to put kids through college and all that crass capitalist stuff. randy --- for your amusement, i can talk about my private play-pen i have a rack in seattle's carrier hotel with 2xSTM-1 connectivity. in it, i have o asterisk running on a freebsd server (many many thanks to the freebsd porting crew) o cisco 1750 with pots out-dial on fixed ld price plan in our home nearby on bainbridge island, we have o cisco 7960 on an external static address o spa-3000 on an external static address - one port to in-house phone system - pstn port to local telco, qwest in our home on the big island of hawaii, we have the same as bainbridge o cisco 7960 on an external static address o spa-3000 on an external static address - one port to in-house phone system - pstn port to local telco, verizon we use the system to get o follow-me forwarding to wherever we are, including mobile phones, blah blah, so that callers don't have to know where we are to call us o free calls between the two houses o low cost calls within north america o gate to low cost voip intl pstn gateway provider, as we make a lot of personal intl calls fairly simple and boring. and thanks to a number of folk who helped me up the learning curve (sjw being the first, and i am not even paying his counselling [sic] bill:-). the telco part of this stuff is not easy for an over-attenuated ip kinda guy. and my programming language background does me no good with asterisk config files! :-) -30- ___ 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] Re: Cisco 7960 Reboots when SoftPhone calls it?
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00 image/firmware, due to a ton of fixes from P0S3-06-3-00 which we were running. But now when I call my phone using X-Lite, the second I answer, it reboots. last eve, i was using xlite in australia - asterisk in states - 7960-7.1 in states for over an hour and it worked fine. you have something strange going on. as it happens for you on multiple 7690s, perhaps sending your 7690 config along would give us a clue? randy ___ 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] Re: SPA-2000, call for help testing echo issues...
with an spa-3000, but spa3k/line1 - asterisk - xten i.e. a pure voip connection, from the states (spa3k) to australia (xten), i heard vicious echo from the states end. the cairns end heard no echo. going to spa3k/line1 - asterisk - sipphone.com - australian/pstn gave no echo. [0] randy --- [0] - though i had to listen to sipphone's enraging commercial when i placed the call. anyone care to recommend a different *international* pstn gateway provider? ___ 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] saydigits/background
is there a way to do SayDigits() or equivalent that is
backgrounded?
application is
exten = s,1,Background(zz-fwd-areyouat) ; use callerid or enter
exten = s,2,SayDigits(${CALLERIDNUM}) ; telling callerid
exten = _*,1,Macro(fwd-set,${userid},${CALLERIDNUM})
exten = _*,2,SayDigits(${CALLERIDNUM}) ; if * use callerid
exten = _*,3,Background(zz-fwd-callswillbe) ; report to where calls
exten = _*,4,SayDigits(${EXTEN}); will be forwarded
exten = _*,5,Hangup()
exten = _X.,1,Macro(fwd-set,${userid},${EXTEN})
exten = _X.,2,Background(zz-fwd-callswillbe); if digits use them
exten = _X.,3,SayDigits(${EXTEN}) ; as outbound number
exten = _X.,4,Hangup()
exten = h,1,Hangup()
exten = i,1,GoTo(s,1)
exten = t,1,GoTo(s,1)
[ yes steve, i spent the usual half hour with google and the
wiki ]
randy
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[Asterisk-Users] Re: Do people actually answer questions here?
we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. randy ___ 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] Re: Do people actually answer questions here?
DO NOT USE GROUP REPLY!!! fat effing chance. fix your mail system. see a very large number of threads. but as you seem unable to look up archives:-), try this in your .procmailrc # prevent dupes # :0 Wh: msgid.lock | formail -D 65536 msgid.cache Booo whh, use better tools. Find out how to use Mozilla and tabbed browsing. been doing that for some years. it can not make up for a bad archive or weak search tools. Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. this is the cult of this is a very complex area. you need to pay us gurus to do it for you. in my 40 years of computing i can not count the fields where i have seen the guru-friendly products and technologies go one of two ways, marginalization and failure or takeover by the massive companies who then marginalize the engineers. i suggest you have another career path planned. If you knew how much time and how much money both of my personal fincances and my employers was put into the knowledge I have currently, you would start to understand why I am so annoyed that you don't seem to want to spend the time it takes to learn. nice of you to turn my comment on the difficulty of the tools into an attack on my knowledge. particularly amusing if you knew more. there are reasons there are so many repeated questions on this list. some of the reasons are weakness in the documentation. we can pretend this is not a problem and attack the questioners or those who try to discuss the weaknesses, or we can discuss how we might address them. clearly you have made your choice. randy ___ 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] Re: Do people actually answer questions here?
From: Steven Critchfield [EMAIL PROTECTED] To: Randy Bush [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] ROFL! ___ 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] Re: Re: Do people actually answer questions here?
Normally I wouldn't respond to such an ignorant thread, but this time I just couldn't resist. As someone who knew absolutely nothing about * a number of months ago, I have to say that I was able to install and configure * the first time in only a matter of hours using nothing more than the information that was available online (either in the wiki or through google searches). Yes, some of the * configuration files can seem daunting at first (it is a steep learning curve), but once you get started and take the time to actually read the configuration files and learn how they actually work the configuration is really not that bad. There have been a few times where I found myself stuck and I found that if I asked a clear, intelligent question either in this list or on IRC, that I would get a clear, intelligent answer. My suggestion to you is, instead of complaining, if you have a problem search for an answer, if you still can't find one, try IRC, if no one there is able to help you out, post a question here. I think you will find that if you continue to search for the answer on your own though, you will end up figuring your problem out without help from anyone else. it is amusing that the people complaining about others not reading the archives etc., seem unable/unwilling to actually read the mail to which they are responding. i did not ask a question. read that again. i did not ask a question. i did triy to suggest that some of the reasons that there were so many questions were the complexity of document navigation. i also suggest that o there will always be a lot of newbie questions o X% of newbies will not check the docs even if they are simple and eloquent o if you're burned out and don't want to answer their questions, look for the D key over at the left hand side of your keyboard o flaming folk will not get them to do better research; it will get them to use worse products o if you have a bit of patience, instead of saying use the effing wiki, idiot, you might try try the following search strategy in the wiki o i.e., as the navigational tools are not up to the complexity, maybe teach navigation randy ___ 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] Re: WaitExten substitute
I never heard of the app WaitExten http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20WaitExten but you could do the following: exten = _.,1,DigitTimeout,5 exten = _.,2,ResponseTimeout,10 exten = _.,3,SetVar(areacode=666) exten = _.,4,Background(zz-in-who) include = extensions include = applications include = speeddials exten = i,1,PlayBack(Invalid-Ext) exten = i,2,HangUp exten = t,1,PlayBack(NoExtensionEntered) exten = t,2,Hangup yes, i could; and i did. problem is that it does not seem to work; hence my posting. it plays the background and then falls through to invalid on the first keypress. i suspect this may be a sipura config issue again causing a double invite; but i am not sure. how do i hack this? Don't, I suggest you read the handbook, the wiki, and/or the archives. been there. done that. but thanks for the pointers. [ btw, search function in wiki is not real great, to be polite; but that issue is not local to this wiki ] randy ___ 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] Re: 7960 straight through?
Anyway, it appears as though the two contexts you have listed below have the exact same name in-internal, sorry, my error in anonymizing the stuff. the dupe is not in the real config. randy ___ 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] Re: 7960 straight through?
my current, inherited, dialplan.xml is DIALTEMPLATE TEMPLATE MATCH=00,1.. Timeout=0 User=Phone / TEMPLATE MATCH=00,* Timeout=5 User=Phone / TEMPLATE MATCH=* Timeout=5 User=Phone / /DIALTEMPLATE the last of the three entries would seem to be the significant one. but my problem is that * is wanting the cisco to prepend its own extension number to the dialed string. see my original message (corrected) below. a ether dump of the sipura's invite shows From: biwa phone sip:[EMAIL PROTECTED];tag=3a553a2b9373c699 To: sip:[EMAIL PROTECTED] ^^^ while cisco demands that i dial the 142 before it will send the invite at all randy --- From: Randy Bush [EMAIL PROTECTED] To: splatters [EMAIL PROTECTED] Subject: 7960 straight through? Date: Thu, 17 Jun 2004 17:42:36 -0700 if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) the problem *** hit Dial then dial 666 sip.conf for crisco [fiji] callerid=crisco 142 type=friend host=dynamic port=5060 secret=pfui qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=in-internal extensions.conf [in-internal] exten = s,1,Answer exten = 141,1,GoTo(int-extns,s,1) ; spa-x000 exten = 142,1,GoTo(int-extns,s,1) ; 7960 [in-extns] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,PlayTones(dial) exten = 141,1,Macro(dial-extension,marais) exten = 142,1,Macro(dial-extension,fiji) exten = 666,1,Macro(dial-extension,downthere) -30- ___ 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] Re: Re: 7960 straight through?
i guess i am not being sufficiently clear.
the sipura seems to act one way and the cisco another. the
sipura, x141, is happily served by
[in-206-sipura]
exten = s,1,SetVar(areacode=206)
exten = 141,1,GoTo(in-int,s,1)
[in-int]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = _001,1,SetVar(mailbox=001)
exten = _001,2,Macro(fwd-call,${EXTEN})
exten = _002,1,SetVar(mailbox=002)
exten = _002,2,Macro(fwd-call,${EXTEN})
i.e. it sends one invite with its own extension and a second
with the dialed extension.
but the cisco requires
[in-206-cisco]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = s,5,SetVar(areacode=206)
exten = _001,1,SetVar(mailbox=001)
exten = _001,2,Macro(fwd-call,${EXTEN})
exten = _002,1,SetVar(mailbox=002)
exten = _002,2,Macro(fwd-call,${EXTEN})
which does not work with the sipura.
either is fine with me. both are not because they require
maintaining a per-extension device-dependent mess in my
sip.conf and two configs in extensions.conf.
randy
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Re: [Asterisk-Users] Re: Re: 7960 straight through?
Ahh. Okay, I think I see. The Cisco isn't doing anything weird; the sipura is. Why is it sending its own extension first? bingo! thank you. fixed. the sipura dialplan was hacked and left in a bad state. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: 7960 straight through?
[in-206-cisco]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = s,5,SetVar(areacode=206)
exten = _001,1,SetVar(mailbox=001)
exten = _001,2,Macro(fwd-call,${EXTEN})
exten = _002,1,SetVar(mailbox=002)
exten = _002,2,Macro(fwd-call,${EXTEN})
What's the point of the 's' extension here?
i really only need the SetVar()
Unless Asterisk does something weird with it that I haven't seen before,
then you'll only get 's' in this context if you get the cisco to dial
without specifying a number.
oops! then how do i get a per-incoming-context SetVar?
randy
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[Asterisk-Users] Re: Re: Re: 7960 straight through?
[inside]
exten = 2000,1,Dial(foo)
exten = 2001,1,Dial(bar)
...
[inside-sip]
exten = _.,1,SetVar(areacode=206)
exten = _.,2,Goto(inside,${EXTEN},1)
doh thanks
randy
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[Asterisk-Users] WaitExten substitute
i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten = _.,1,SetVar(areacode=666) exten = _.,2,Background(zz-in-who) ; give them list of extns exten = _.,3,WaitExten(10) ; let them enter extn to call include = extensions include = applications include = speeddials exten = i,1,HangUp exten = t,1,HangUp how do i hack this? randy ___ 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] dialtone stop
this must be a very faq, but wiki and google are not yielding
much except for the appended message.
i want the tone, which i am currently producing with
PlayTones(dial) to stop when the caller hits a key.
as in
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
include = parkedcalls
exten = _42,1,SetVar(mailbox=001)
exten = _42,2,Macro(fwd-call,${EXTEN})
randy
---
Subject: [Asterisk-Users] new application Dialtone()
Date: Wed, 4 Jun 2003 16:16:49 -0700
From: Surfer Dude [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Organization: Swell, Inc.
To: [EMAIL PROTECTED]
Hello,
I created a new application for myself called Dialtone() by modifing
res/res_indications.c file. It can be used as such:
exten = s,4,Dialtone(30|${CALLERIDNUM})
exten = s,5,Playback(time-exceeded)
exten = s,6,Goto(s|1)
It will stutter if you have new voicemail and you have passed the mailbox
number as I did above. It will stop dialtone the moment you press a key or
the timeout (in seconds) has occured.
This is my first stab at asterisk so I am not 100% that this is where the
code should live. If there is someone out there that wants to add it to (*)
then please feel free. I can send you the whole file if you need.
I needed this application because I need to set immediate=yes so that I can
set some variables when the phone is picked up.
Thanks,
Jason
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[Asterisk-Users] 7960 straight through?
if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid=crisco 142 type=friend host=dynamic port=5060 secret=pfui qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=in-internal extensions.conf [in-internal] exten = s,1,Answer exten = 141,1,GoTo(int_extns,s,1) ; spa-x000 exten = 142,1,GoTo(int_extns,s,1) ; 7960 [in-internal] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,PlayTones(dial) exten = 141,1,Macro(dial-extension,marais) exten = 142,1,Macro(dial-extension,fiji) exten = 666,1,Macro(dial-extension,downthere) randy ___ 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] Re: Asterisk-Users List Etiquette
Before hanging up, there should be an extension reminding everyone that top posting is super duper wrong and oh so annoying. I used to top post, but now I understand why it's frowned upon. It should be avoided at all costs. Unfortunately, some of us just can't help it. Some of us use this nifty (read: *$#[EMAIL PROTECTED]) program called Outlook (especially in the workplace), made by a company that doesn't care for common email ettiquete, let alone user preference. I happen to be one of those persons (although it's not a matter of the company's IT decisions; I just don't see how it is beneficial to use multiple emailers for the sake of a mailing list. And for those that *are* subscribed at the workplace, note that it took a bit of effort (too much to write a simple message like this, IMO) to produce a properly bottom posted message like this (that is, if you consider the attribution without date proper, which would require more copying, pasting, and lead to further obfuscation). And for those of you who don't like HTML email with different fonts or colors, etc., there's this thing called CSS. If you expect everyone else to use a client that bottom-posts, then I expect you to use an HTML-capable email client that supports CSS for accessibility. Yes, I hear your reply, and you're right, switching clients is not an answer for everyone. Deal with it. thanks for your non-html and bottom posted message randy ___ 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] inviting an spa-x000
sip debug shows that my * is trying to invite my spa and being told 404 Reliably Transmitting: OPTIONS sip:42.7.11.194 SIP/2.0 Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7 From: asterisk sip:[EMAIL PROTECTED];tag=as39d40d19 To: sip:42.7.11.194 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 14 Jun 2004 23:13:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 42.7.11.194:5060 Sip read: SIP/2.0 404 Not Found To: sip:42.7.11.194 From: asterisk sip:[EMAIL PROTECTED];tag=as39d40d19 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7 Server: Sipura/SPA1000-2.0.8(GW) Content-Length: 0 the sip.conf entry for the spa is [spa0p] callerid=SPA Phone 102 type=friend host=dynamic port=5060 username=spa0p auth=md5 secret=seekret qualify=1000 dtmfmode=rfc2833 canreinvite=no context=in-99 mailbox=001 it appears that * is attemting to authenticate as 'asterisk'. but then what is the username in the sip.conf entry? randy ___ 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] Re: Local calls to x100p all else to iax term
I have a list of all my local prefixes(free) on my POTS. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX ; your first prefix will be 555 exten = _91831555,1,Dial(pstn...) ; your second prefix will be 678 exten = _91831678,1,Dial(pstn...) ; ok, no match from any of the previous 60 prefixes, so IAX exten = _91NX.,1,Dial(IAX...) Remember that you can combine prefixes using the exten regular expression: ; your first prefix will be 555, 556, 557, 558 or 559 exten = _9183155[5-9],1,Dial(pstn...) i think the point may be that there can be massive prefix lists and folk don't want the extensions.conf from hell. it would be a nice hack to have an external utility to load one's named [0] prefix lists into something that can be used for matching analogously to DBGet(). randy --- [0] - named so one can have more than one. i have out-dials in four locations each with a non-trivial set of prefixes. so i kinda wanna do exten = biwa.,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = hawi.,2,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = marais.,3,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = tokyo.,4,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = _.,5,Dial(SIP/[EMAIL PROTECTED],60,Ttr) and let some external process/cron/gui/... keep the four databases up to date. ___ 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] Re: Re: Local calls to x100p all else to iax term
include file = prefixes.conf kinda. but i would prefer to separate the list of prefixes from the dialplan policy of how to deal with them. randy ___ 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] Re: Asterisk on FreeBSD News
Asterisk on FreeBSD News thank you! ___ 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] Re: Cisco Auto Provisioning
However, the task is to create config files for each device and place them in TPTP root directory and apparently one needs cfgmfg and pdat files in order to create these config files. man tftpd and look for -C ___ 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] Re: Re: NetworkWorld article on Open Source Telephony
Obviously, you have seen very few OM interfaces. I happen to feel that Cisco IOS is the most beautifull inteface known to present day man... The power of asterisk comes from its method of config. yup. it meets the challenge of finding something more complex, less intuitive, less parsable, and less managable than crisco ios. another exceedingly incorrect assumption randy --- Q: Because it reverses the logical flow of conversation. A: Why is top posting frowned upon? ___ 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] Re: NetworkWorld article on Open Source Telephony
It's more the general day to day maintenance that needs to be addressed depends on whether your more senior geeks have a desperate need for job security. maybe they like adding users, extensions, ... as opposed to letting administrative types do it. randy ___ 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] Re: Re: DNS SRV records
Time for Duane to start implementing DNS SRV, since it's from now on is turned on by default in CVS head. Unless you're planning on breaking other standards my A records will keep on working just fine :) except you (likely to be ex-) customers will have problems reaching more and more of the universe. as the idiom goes, not a problem to me. randy ___ 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] Re: NetworkWorld article on Open Source Telephony
The power of asterisk comes from its method of config. yup. it meets the challenge of finding something more complex, less intuitive, less parsable, and less managable than crisco ios. randy ___ 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] Re: Re: NetworkWorld article on Open Source Telephony
I happen to feel that Cisco IOS is the most beautifull inteface known to present day man... women know better. get a shrink. or better yet, take a compiler 101 course. randy ___ 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] Re: Re: Re: DNS SRV records
Yes I've been told how much you love enum a few hints o i was one of the gang who came up with the idea of stealing the enum hack from tcp.int (may have been the first, but the brain, such as it is, fades) o i helped the sippers work out the enum naptr etc hacks, in fact pushed them from cname to naptr o i asked cisco for support of domain-free enum many years ago (and did not get it) o blah blah blah so, you are welcome. but yes, i did come to realize that mapping phone numbers was pretty silly; the internet has its own ways of naming things, and the telcos' is s old and broken, technically and socially. so i could care less how many mee-toos wanna map phone numbers into their own domains; it all seems a bit silly from my pov. but enjoy. which all as nothing to do with the issue at hand, proper support of srv and naptr rrs in asterisk. and enum needs 'em too. so, as olle has said, gotta happen. randy, looking at the code and for spare time ___ 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] Re: DNS SRV records
Exactly my point, by ***DEFAULT*** Asterisk won't use SRV records, is this a feature or a bug? randy ___ 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] sip call using name in sip.conf
i try to place a call exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid=local ext 103 1914666 type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \ No such host: foo May 22 23:11:31 NOTICE[140400128]: app_dial.c:536 dial_exec: \ Unable to create channel of type 'SIP' the sip service is registered foo/foo 209.20.186.194 (D) 255.255.255.255 5060 Unmonitored and i get the same result if it is not dynamic foo/foo 209.20.186.194 255.255.255.255 5061 Unmonitored clues appreciated randy ___ 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] sip call using name in sip.conf
[foo] type=friend I do not beleive that will work for type=friend. If you use separate type=peer and type=user blocks in sip.conf it may work. Expecially if you also specify a port in the Dial(). Else, use the hostname (or a const). hmmm. then, how do i let it be dynamic if it has two blocks in sip.conf, one for inbound and one for out? i.e, how does it register its ip address in both? randy ___ 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] Re: Asterisk on Compact PCI platform
Good call -- write cycle life of 10^3-10^4 are probably not much of an issue in a digital camera, but would probably die quickly if used as a HD replacement. i have a cigarette-sized freebad box using only flash for disk and swap. runs for years under load. a bunch of us use them, though the rest of the folk are netbsd deviants:-) . randy ___ 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] avoiding rtp triangle
so, is it safe to put canreinvite=yes on a 7960? on a 1750? on a spa-x000? an xten? how the heck do i find out other than the hard way? randy -- ps: pun intended ___ 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] registering in sipphone
for inbound calls, i can register
context = from-sipphone
register = 1747xxx:[EMAIL PROTECTED]
but how do i configure to make outbound calls to them?
exten = _1747XXX,1,GoTo(dial-sipphone,${EXTEN},1)
[dial-sipphone]
;
; SIP to sipphone.com
;
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
^^
exten = _X.,2,Playtones(congestion)
exten = _X.,102,Playtones(busy)
exten = h,1,Hangup
randy
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[Asterisk-Users] want to set a var in sip.conf
i have extensions in locations across a number of telco area codes. when someone in seattle picks up and dials 91234567, it would be nice to transform it to 92061234567. i would prefer not to have an extension context per area code. it would be cool to be able to set a variable in the sip.conf bit for each phone with it's geographic default area code. or other folk may have a better hack. please clue me. the wiki did not help. thanks. randy ___ 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] vovidia
so, on 23 april, i ordered a sipura 3000 via vovidia. they charged my credit card and delivered nothing. shall i raise a fuss, or is this considered normal? randy ___ 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] OMG THE SKY IS FALLING!! NOT!!!
while it is true that there are other insecure systems and protocols, we should not use that as an excuse for the fact that sip/rtp are pretty damned insecure. and the current ietf approach is to complicate them so much that no one will ever be able to have any confidence that an implementation is reasonably secure or not. randy ___ 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] Cisco 7960 SIP Firmware
you have sent a message to me which seems to contain a legal warning on who can read it, or how it may be distributed, or whether it may be archived, etc. i do not accept such email, and have therefore deleted it. do not expect further response. randy ___ 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] list batching frequency
subscribers to the digest form of this list do so in order to only receive the email infrequently. in my case, and i suspect others, twice or so a day would be preferred. the list currently batches about every hour. it is sufficiently annoying that one tends to delete batches. i have written to the list admin about this and received no response, undoubtly they are busy reading the mail :-). would anyone reading the digest form object to the admin changing the config so it sends one to three batches a day? randy ___ 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] freebsd?
make install -DNO_IGNORE h, scary considering i don't need h323. or am i misunderstanding something? I'm also working on a freebsd port that uses the cvs version of asterisk, let me know if you're interested in taking a look. o! but i am about to go back on the road. so i don't know if i will have time this week. randy ___ 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] freebsd?
the freebsd port tree version is dead because of the openh323 issues. before i start hacking, i am hoping someone else has a freebsd version that will build on -current. and i do not care about h232. dare i hope? randy ___ 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
