Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13
I guess the problem is that * sends the response to port 5060, while the phone listens on port 2xxx for an answer. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White Gesendet: Montag, 29. Oktober 2007 07:46 An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13 Here are more details: The phone and the Asterisk box are behind the same router (the Asterisk machine is 192.168.0.2 and the phone is 192.168.0.4). A ping command works: [EMAIL PROTECTED]:~$ ping -c 10 192.168.0.4 PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data. 64 bytes from 192.168.0.4: icmp_seq=1 ttl=64 time=0.500 ms 64 bytes from 192.168.0.4: icmp_seq=2 ttl=64 time=0.491 ms 64 bytes from 192.168.0.4: icmp_seq=3 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=4 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=5 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=6 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=7 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=8 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=9 ttl=64 time=0.505 ms 64 bytes from 192.168.0.4: icmp_seq=10 ttl=64 time=0.492 ms --- 192.168.0.4 ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9005ms rtt min/avg/max/mdev = 0.491/0.495/0.505/0.014 ms [EMAIL PROTECTED]:~$ However, the phone never appears to receive the responses from Asterisk to its register requests. The error on the phone is: [2]29/10/2007 17:02:59: Transport Error: Pending packet 1046807: generating fake [2]29/10/2007 17:02:59: Registrar [EMAIL PROTECTED] timed out From /etc/asterisk/sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to A realm=192.168.0.2 context = default ;Default for incoming calls [5549] disallow=all allow=ulaw allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) secret=localphone ; obvious password for testing host=dynamic callerid=Jason White 5549 dtmfmode=auto mailbox=5549 ;(Asterisk VM-system's mailbox #) The output from sip set debug is attached, as captured earlier by the script command. Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this phone has successfully registered with external Asterisk servers. Suggestions are much appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP multi Bindport
Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Thank You Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13
On Mon, Oct 29, 2007 at 08:17:20AM +0100, Christian Stredicke wrote: I guess the problem is that * sends the response to port 5060, while the phone listens on port 2xxx for an answer. That could be the problem. The phone specifies port 2048 in its contact field. Is there a way to configure Asterisk to respond on whichever port the phone specifies? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320 phonewithAsterisk 1.4.13
Well, the response should go to the port number provided in the Via header. If there is a rport set, then to that port. Everything looks good in the log, the only problem is that the response is sent to the wrong port. The Contact port will be used later when the server wants to send a request (not a response) to the phone. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White Gesendet: Montag, 29. Oktober 2007 09:16 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Registration of Snom 320 phonewithAsterisk 1.4.13 On Mon, Oct 29, 2007 at 08:17:20AM +0100, Christian Stredicke wrote: I guess the problem is that * sends the response to port 5060, while the phone listens on port 2xxx for an answer. That could be the problem. The phone specifies port 2048 in its contact field. Is there a way to configure Asterisk to respond on whichever port the phone specifies? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320 phonewithAsterisk 1.4.13
On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote: Well, the response should go to the port number provided in the Via header. If there is a rport set, then to that port. Everything looks good in the log, the only problem is that the response is sent to the wrong port. I tried inserting nat=never into sip.conf but that didn't help. Is there a configuration option that will fix this? If not, what's the prospect of having it corrected for the next release of Asterisk? I can test a patch if that would help. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
What you can still to is setting the port on the phone to port 5060 - just as a little dirty workaround until there is a better solution available. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White Gesendet: Montag, 29. Oktober 2007 10:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13 On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote: Well, the response should go to the port number provided in the Via header. If there is a rport set, then to that port. Everything looks good in the log, the only problem is that the response is sent to the wrong port. I tried inserting nat=never into sip.conf but that didn't help. Is there a configuration option that will fix this? If not, what's the prospect of having it corrected for the next release of Asterisk? I can test a patch if that would help. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
On 10/29/07, Christian Stredicke [EMAIL PROTECTED] wrote: What you can still to is setting the port on the phone to port 5060 - just as a little dirty workaround until there is a better solution available. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White Gesendet: Montag, 29. Oktober 2007 10:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13 On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote: Well, the response should go to the port number provided in the Via header. If there is a rport set, then to that port. Everything looks good in the log, the only problem is that the response is sent to the wrong port. I tried inserting nat=never into sip.conf but that didn't help. Is there a configuration option that will fix this? If not, what's the prospect of having it corrected for the next release of Asterisk? I can test a patch if that would help. snom phones have been using ports in the 2000+ range since the dawn of asterisk without any problems, so I suspect that this will be an Asterisk configuration error, or a change to the asterisk SIP stack that is causing problems. Can you also check that the snom has a suitably recent firmware version. It may be a bug in something the phone is sending. On the other hand, changing the port number on the phone might be the quickest solution :) Cheers, Steve Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI span configuration - span remains down
Just a little follow up here... Missed a call from someone at Telewest on friday, so I don't know what they were going to tell me. However I've come in this morning and thought well, you never know, perhaps he was phoning to say we've fixed it. Tried calling my mobile and now it works. No idea what was done, but it wasn't by me. I guess if any of you live in England, you might be nodding and going, ah, yes, NTL/Telewest... :) Cheers for suggestions and help Dave. On 10/26/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 10/26/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Is there some part of the debug output I need to tell the telco about? When I was on to them earlier today, the engineer only seemed to know how to turn bits of their network on and off, not much about settings I need my end etc. Just tell them when you try to make a call, you get cause code 44 back (channel unavailable). They can look at their switch to figure out what's going on. I had a strange problem with cause code 44 on just 5 B channels of a PRI. The first time I'd dial, I'd get cause code 44 and * would attempt to restart the B channel. The switch would never respond to the request to restart, so the channel remained in limbo from *'s perspective, and further attempts to dial out explicitly on that channel would give me congestion (generated from *, not from the Telco), and attempts to dial out using a group that contained those channels would just skip over them. I called the Telco, and spent over a week trying to convince them that the RELEASE COMPLETE was coming from their end. They claimed it was coming from me. It was almost as if something in between my system and where the tech was running his trace was proxying the Q.931 messages, and sending us both a cause code 44 when I used those channels. In the end, they re-built my trunk and the problems immediately cleared, so it was apparantly some buggered state in their switch. This was with a 5ESS running NI-2 if that helps. -- j. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
On Mon, Oct 29, 2007 at 10:19:57AM +0100, Christian Stredicke wrote: What you can still to is setting the port on the phone to port 5060 - just as a little dirty workaround until there is a better solution available. Would that be the sip_port settings entry? It is documented as for internal use, though I suppose it shouldn't cause any harm if I change it. Incidentally, this problem may have been addressed in the development sources. Perhaps I should obtain and build an svn checkout. From the svn log: Revision 77616 Modified Sat Jul 28 07:44:16 2007 UTC (3 months ago) by rizzo File length: 681368 byte(s) Diff to previous 77538 make use of received= and rport= fields in sip replies. In a nutshell, these fields are used to tell a sip entity the address and port its request came from, and are extremely useful in the presence of NATs, especially with symmetric NATs where STUN is totally ineffective. This patch stores the address and port in the 'ourip' field of the dialog descriptor, so they can be reused in subsequent transactions. As it is, it works well for things like REGISTER requiring authentication, because the second REGISTER request (with auth credentials) will carry the correct address. Maybe it can also be useful, in case of an address change, to do one or both of the following: + propagate the new address to the parent user/peer descriptor so that new dialogs will use the correct address from the beginning. This is trivial to implement, I am just waiting for feedback on this. + re-issue a request in case of an address change. This a lot less trivial, maybe unnecessary, and probably covered by the previous item. I would seriously consider this patch for addition to 1.4 and 1.2. The code is very little intrusive, and it would solve in a correct way the nat traversal problems for which externip/externaddr/stunaddr are only a partial and expensive workaround. __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use instead of LookupCIDName?
On Thu, Oct 25, 2007 at 07:13:52PM +0200, Vincent wrote:
On Thu, 25 Oct 2007 18:46:19 +0200, Vincent
[EMAIL PROTECTED] wrote:
I guess I should use this as a parameter to a function, but which one?
Never mind, I found how to use it:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
Yes - not deprecated until 1.4 of course but it even works in 1.2, I
have already moved over to it.
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o mail: [EMAIL PROTECTED]
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[asterisk-users] Realtime context
Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include = parkedcalls switch = Realtime/@ [fromiax] switch = Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch = Realtime/@ but I have about 50 asterisk that read one database, now if I want to change/add a context I have to change 50 extensions.conf file :( Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] issues with downloads.digium.com
Hi Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED] have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at http://downloads.digium.com/pub/telephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Furthermore, I cannot follow links directly. Links are redirections. For instance, the link marked with aadk points to: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk $ HEAD 'http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk' 200 OK Connection: close Date: Mon, 29 Oct 2007 10:05:54 GMT Accept-Ranges: bytes ETag: 26cb96-963-433e597412940 Server: Apache/2.2.3 (Debian) PHP/5.2.0-8+etch7 mod_ssl/2.2.3 OpenSSL/0.9.8c Content-Length: 2403 Content-Type: text/html; charset=UTF-8 Last-Modified: Wed, 27 Jun 2007 16:18:05 GMT Client-Date: Mon, 29 Oct 2007 10:05:49 GMT Client-Peer: 216.207.245.16:80 Client-Response-Num: 1 As you can see, that script doesn't really redirect. It does not point me to the new file name. If I use a web browser, I still get the illusion of a directory tree, but this breaks any decent attempt of mirroring downloads.d.o . It also breaks downloads with wget. There is no proper date for files as well. A casual look on the files in the directory can no longer tell you when the version was released. Even worse, you cannot use wget -c to avoid a duplicate download, as on the second time you try to download, you have a newer version of the original. Another problem is that an incorrect link will not return an 404 page. It will redirect you to the homepage. In short, HTTP may return some codes other than 200. Try: http://www.joek.com/404 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP multi Bindport
you can do it using iptables, port forwarding. On 10/29/07, Abdul [EMAIL PROTECTED] wrote: Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Thank You Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
Tzafrir Cohen wrote: Hi Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED] have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at http://downloads.digium.com/pub/telephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Furthermore, I cannot follow links directly. Links are redirections. snip Not sure I completely understand what you mean by I have to guess which of them is a file and which is a directory. When I open any location in downloads.digium.com in either firefox or konqueror I can see that directories have a / after their name and also use a folder icon. That aside, I don't care for the usage of the redirect scripts either. I used to be able right-click and copy the URL and paste it into wget (browse on one machine and download on another). You can still type the correct URL into wget and perform the download, but that's sooo many more keystrokes. Lynx doesn't seem to work at all and with Links I have to answer Accept to Java script is attempting to to go URL every time I select a link. Drag and drop copying with Konquerer doesn't work either. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO ATA Options?
Conall O'Brien wrote: Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? I use one at home and can recommend it as functional and reliable. It has an unbelievable number of configuration options. Linksys docs are a bit sparse, try the Sipura site under SPA3000. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDI setup help
HELLO ALL!
I followed a tutorial called DUNDi so easy to set up DUNDi peers.
Unsurprising it was not that easy hehe.
I have the following files up and running, peers are visible but when I
do a query e.g dundi lookup [EMAIL PROTECTED] I get the following error.
CAUSE: NOAUTH: Unsupported DUNDi context.
Could anybody help?
Thank you kindly.
James
=== IAX.CONF on both servers ===
[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi
=== DUNDI.CONF FILE ON SERVER 1 ===
[mappings]
priv=dundiextens,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},noparti
al
[00:0B:CD:08:23:00] ;We can see server .151
model=symmetric
host=XXX.XXX.XX.151
inkey=dundi
outkey=dundi
include=priv
permit=priv
qualify=yes
order=primary
=== DUNDI.CONF FILE ON SERVER 2 ===
[mappings]
priv=dundiextens,0,IAX2,priv:${SECRET}XXX.XXX.XX.151/${NUMBER},nopartia
l
[00:0B:CD:08:22:F6] ;We Can see the Server .150
model=symmetric
host=XXX.XXX.XX.150
inkey=dundi
outkey=dundi
include=priv
permit=priv
qualify=yes
order=primary
=== DUNDI.CONF FILE ON SERVER 1 ===
[General]
[lookupdundi]
;this is where DUNDi querys the peers and requests and extension
switch = DUNDI/priv
[dundiextens]
;this is where we list the actual extensions that this pbx responds to
exten = AS,1,NoOp
[incomingdundi]
;this is the entry point where dundi calls come into this server - we
specified this context in iax.conf
;simply forward the actual extenstion into [internal] using goto
exten = AS,1,Goto(internal|AS|1)
[internal]
;change the context and executethe switch statement which enables dundi
to query the peers
include = lookupdundi
; phone line AS
exten = AS,1,MixMonitor(ASDUNDI.wav|av(0)V(0))
exten = AS,2,Dial(SIP/ASCHCP)
exten = AS,3,Answer()
exten = AS,4,Busy(10)
exten = AS,5,Hangup()
=== DUNDI.CONF FILE ON SERVER 2 ===
[General]
[lookupdundi]
;this is where DUNDi querys the peers and requests and extension
switch = DUNDI/priv
[dundiextens]
;this is where we list the actual extensions that this pbx responds to
exten = DO,1,NoOp
[incomingdundi]
;this is the entry point where dundi calls come into this server - we
specified this context in iax.conf
;simply forward the actual extenstion into [internal] using goto
exten = DO,1,Goto(internal|DO|1)
[internal]
;change the context and executethe switch statement which enables dundi
to query the peers
include = lookupdundi
; phone line DO
exten = DO,1,MixMonitor(DODUNDI.wav|av(0)V(0))
exten = DO,2,Dial(SIP/DO)
exten = DO,3,Answer()
exten = DO,4,Busy(10)
exten = DO,5,Hangup()
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Re: [asterisk-users] issues with downloads.digium.com
In article [EMAIL PROTECTED], Tzafrir Cohen [EMAIL PROTECTED] wrote: Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED] have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at http://downloads.digium.com/pub/telephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Furthermore, I cannot follow links directly. Links are redirections. For instance, the link marked with aadk points to: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk $ HEAD 'http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk' 200 OK Connection: close Date: Mon, 29 Oct 2007 10:05:54 GMT Accept-Ranges: bytes ETag: 26cb96-963-433e597412940 Server: Apache/2.2.3 (Debian) PHP/5.2.0-8+etch7 mod_ssl/2.2.3 OpenSSL/0.9.8c Content-Length: 2403 Content-Type: text/html; charset=UTF-8 Last-Modified: Wed, 27 Jun 2007 16:18:05 GMT Client-Date: Mon, 29 Oct 2007 10:05:49 GMT Client-Peer: 216.207.245.16:80 Client-Response-Num: 1 As you can see, that script doesn't really redirect. It does not point me to the new file name. If I use a web browser, I still get the illusion of a directory tree, If you get the body too, you will see that it is actually an HTML page with a lot of embedded JavaScript. It is the JavaScript that makes the browser load the target page. It appears to be part of a package from a company called Eloqua (www.eloqua.com). but this breaks any decent attempt of mirroring downloads.d.o . It also breaks downloads with wget. I suspect that this is an intentional design decision in the Eloqua package, and probably desired by some of Eloqua's customers. Whether it is an appropriate package to be used by Digium to serve open-source content is another matter altogether Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO ATA Options?
Hi, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. i've had some problems setting the disconnect tone correctly to my country. As a matter of fact, i still do, as the calculated values does not always hang up the phone. Other than this i have a small issue which i did not understand completely. Sometimes the SPA webpage starts to load, and in the middle i get a connection reset page by firefox. Sometimes i can only load the page by refreshing 10-15 times or even more. This only happens in advanced admin mode, when using any other modes everything works fine. This refresh-error only occures from remote networks, not from a PC that is within the same subnet as the SPA (subnets are connected via site2site vpn tunnel). I haven't had time to correctly debug this issue (tcpdump etc) but it's so annoying that i will go and debug this once. It may be an MTU issue, an SPA performance issue, a firefox issue... This is an SPA3k which i'm using (actually not one but four, all involved in this problem). regards Adam ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Dialplan] Actions
Hello I'm learning more about dialplans and have a couple of questions: 1. Am I right in understanding that the actions that can be performed in extensions.conf can be of two types only: - internal commands (Dial, Wait, etc.) - calls to external scripts throught AGI? 2. I'd rather write scripts in Python instead of Perl or PHP. Does someone have a skeleton that I could use to build one, including how to call it from extensions.conf with parameters (from * to script, and from script back to *)? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
Tzafrir Cohen wrote: Furthermore, I cannot follow links directly. Links are redirections. For instance, the link marked with aadk points to: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk $ HEAD 'http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/aadk' 200 OK Connection: close Date: Mon, 29 Oct 2007 10:05:54 GMT Accept-Ranges: bytes ETag: 26cb96-963-433e597412940 Server: Apache/2.2.3 (Debian) PHP/5.2.0-8+etch7 mod_ssl/2.2.3 OpenSSL/0.9.8c Content-Length: 2403 Content-Type: text/html; charset=UTF-8 Last-Modified: Wed, 27 Jun 2007 16:18:05 GMT Client-Date: Mon, 29 Oct 2007 10:05:49 GMT Client-Peer: 216.207.245.16:80 Client-Response-Num: 1 As you can see, that script doesn't really redirect. It does not point me to the new file name. If I use a web browser, I still get the illusion of a directory tree, but this breaks any decent attempt of mirroring downloads.d.o . It also breaks downloads with wget. In short, HTTP may return some codes other than 200. You are absolutely right. I don't even get a directory listing, maybe because I block most of the Eloqua stuff. Having to copypaste the correct URL every time is just plain annoying. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] XML file for spa devices
Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
Dave Fullerton wrote: Tzafrir Cohen wrote: Let's look at http://downloads.digium.com/pub/telephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Not sure I completely understand what you mean by I have to guess which of them is a file and which is a directory. When I open any location in downloads.digium.com in either firefox or konqueror I can see that directories have a / after their name and also use a folder icon. http://downloads.digium.com/pub/telephony/ : ---cut--- Index of /pub/telephony/ Icon Name Last modified Size Description[DIR] Parent Directory [ ] aadk/ 10-Aug-2007 08:40- [ ] asterisk/ 16-Oct-2007 18:25- [ ] codec_g729/12-Jun-2007 13:20- [ ] firmware/ 02-Feb-2007 10:30- [ ] gastman/ 04-Dec-2005 18:22- [ ] gnophone/ 04-Dec-2005 18:22- [ ] hpec/ 05-Feb-2007 15:15- [ ] libiax/04-Dec-2005 18:21- [ ] libpri/27-Oct-2007 15:10- [ ] sounds/21-Feb-2007 18:40- [ ] zaptel/27-Oct-2007 15:05- [ ] README.contents21-Apr-2007 08:25 625 ---cut--- Icons are fine. README.contents doesn't sort correctly. http://downloads.digium.com/pub/telephony (without the trailing slash): ---cut--- Index of /pub/telephony Icon Name Last modified Size Description[DIR] Parent Directory [ ] README.contents31-Dec-1969 18:00 [ ] aadk 31-Dec-1969 18:00 [ ] asterisk 31-Dec-1969 18:00 [ ] codec_g729 31-Dec-1969 18:00 [ ] firmware 31-Dec-1969 18:00 [ ] gastman31-Dec-1969 18:00 [ ] gnophone 31-Dec-1969 18:00 [ ] hpec 31-Dec-1969 18:00 [ ] libiax 31-Dec-1969 18:00 [ ] libpri 31-Dec-1969 18:00 [ ] sounds 31-Dec-1969 18:00 [ ] zaptel 31-Dec-1969 18:00 ---cut--- Icons are just ?. mtime is wrong. Links do not have a trailing slash. But README.contents sorts correct. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
On Mon, Oct 29, 2007 at 09:02:14AM -0400, Dave Fullerton wrote: Tzafrir Cohen wrote: Hi Sorry to use this public place, but IRC and emails to [EMAIL PROTECTED] have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at http://downloads.digium.com/pub/telephony/ I get a list of items. I have to guess which of them is a file and which is a directory. There is no proper date of change. Furthermore, I cannot follow links directly. Links are redirections. snip Not sure I completely understand what you mean by I have to guess which of them is a file and which is a directory. When I open any location in downloads.digium.com in either firefox or konqueror I can see that directories have a / after their name and also use a folder icon. I later noticed that this seems to be a strange bug (feature?) - if you get to a directory listing URL and don't add the trailing / , you'll get the bogus page I describes. If you do add that /, you'll get a nicer page. http://downloads.digium.com/pub/telephony http://downloads.digium.com/pub/telephony/ Either those are two display modes, or this is a bug. With such an obfuscation script, I can't really tell. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
Rizwan Hisham wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. For SPA-921 and SPA-941, you can get it from the phone itself: http://phoneip/admin/spacfg.xml I'm sure the same goes for SPA-961, but I don't have any of those. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fetch call
Hi, I have asterisk installed. When a connection comes from the outside one of our phones rings for about 45 seconds. Is it possible to another phone fetch the call while it's ringing on the first phone? I don't want to use ringgroups because the second phone would be ringing also. Thanks Nuno Fernandes signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-841 vs Grandstream GXP-2000
I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line, I did pick up one GXP-2000 when it finally went on sale, and it's sort of the odd bird of my office. I haven't been too thrilled with it. Is there anyone out there who honestly prefers the GXP-2000 to the SPA-841? Am I missing some great functionality? I just find the UI to be quirky and cumbersome, though the phone itself is decent. I may just be too used to the SPA way of doing things. I've upgraded the GXP's firmware a few times, but I stopped because it seemed like each upgrade had a step backwards too. Maybe I can find someone out there who wants to trade my GXP for an SPA so I can have a homogeneous phone system. ;) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk: No Longer Answering Calls
Hi Friends! I need help! I'm still Asterisk rookie, so please forgive me. My Asterisk is no longer answering incoming call on the phone line. I call the phone and it rings but asterisk is not picking it up. The phone line is attached to port 4 (FXO) on my digium TDM411P card. I am running Asterisk 1.4.11 with zaptel-1.4.5.1 and libpri-1.4.1 on Fedora Core 5, Linux Kernel 2.6.15-1.2054_FC5smp when I look in the asterisk messages log file I see this interesting messages: - [Oct 29 12:03:09] WARNING[2073] config.c: Unterminated comment detected beginning on line 28 [Oct 29 12:03:09] WARNING[2073] chan_zap.c: Unable to specify channel 4: No such device or address [Oct 29 12:03:09] ERROR[2073] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 [Oct 29 12:03:09] ERROR[2073] chan_zap.c: Unable to register channel '4' [Oct 29 12:03:09] WARNING[2073] config.c: Unterminated comment detected beginning on line 27 - What's baffling about this to me is that Asterisk has answered calls on the same channel before and executed my test dial plan. But now, it is no longer answering calls. Anyone knows what I'm missing here? Also, when I call the box and it is ringing, I connect to the CLI and tell it answer the call, it says No one is calling us but the phone is ringing. The modules are loaded fine. When I check with zAny clues? Thanks, Jeng Following are my very basic test config files, straight out of the Asterisk book: # #/etc/zaptel.conf # # # Define FXO port with FXS signaling # fxsks=4 loadzone=uk defaultzone=uk # # -- End /etc/zaptel.conf --- ;/etc/asterisk/zapata.conf ;- ; [trunkgroups] ; Define any trunk groups here [channels] ; Hardware channels we use ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes ; define channels context=incoming; Incoming calls go to [incoming] in extensions.conf signalling=fxs_ks ; Use FXS signalling for an FXO channel channel = 4; PSTN attached to port 4 ; ;-- End /etc/asterisk/zapata.conf- ; ;/etc/asterisk/extensions.conf ;- ; [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Playback(hello-world) exten = s,3,Background(vm-enter-num-to-call) exten = s,4,Hangup() ; ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
i have spa 2100. tried to access the file but got 404 not found. Any clues why? On 10/29/07, Per Jessen [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. For SPA-921 and SPA-941, you can get it from the phone itself: http://phoneip/admin/spacfg.xml I'm sure the same goes for SPA-961, but I don't have any of those. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP multi Bindport
Am Montag, den 29.10.2007, 15:54 + schrieb Gordon Henderson: On Mon, 29 Oct 2007, Abdul wrote: Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Really? What country?? What ISP? This isn't my experience in the UK .. (yet???) Hi Gordon, I have heard of SIP/VoIP port blocking in certain Asian regions. I think in India the phone market regulations are in favour of their local Ma Bell company who wants to sell minutes, not transport cheap VoIP packets over DSL. I think I read about one of the states of the Arabic peninsula that their jurisdiction forbade any kind of communication that might be considered encrypted or untraceable. Tracing SIP is considered more difficult than wiretapping an analogue line copper pair, figures. I have been told by a friend of mine whose husband-to-be is in Shanghai for a few weeks that VoIP is not restricted there - contrary to the common assumption that the Chinese digital wall is airtight. There might exist restricions in Internet access in rural areas, or for locals (opposed to foreign tourists and workers). There are other regions and legislatures that might prefer strong control of international communications (not necessarily those called Axis of Evil). When I was a child, most of the letters I got from my eastern aunt were inspected, and older locals know of line noises from technologically outdated wiretapping equipment used by the Stasi- might be legends though. I once visited her, crossing the Iron curtain was an intimidating experience for a young boy, even with his father at his side (although other things of the then-East German Republic stuck more in my mind). I am quite glad we can mostly say publically what we think appropriate nowadays. Locals of those countries concerned will know better than me, possibly they are not interested in Asterisk though because of the obvious (legally or technically mandated) uselessness. You might check where those people asking about OpenVPN/Sip combination are from ;-) fiction If and once the more restrictive politicians take control and realize their personal idea of 1984, you surely will also notice the telecommunications regulations, that according to MINITRUTH, will have been there all the time. I am positive they will also cover the airstrip one region of Oceania, so don't run, they will come for you. (Orwell's 1984 was one of my final exam topics at high school) /fiction Coming back to reality I wish you a nice evening. Best regards Anselm *Wait, there's someone on the door, I !%$§)(A/SCNR!)(/§ CARRIER LOST ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A Leg Control on Asterisk Callback
I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP multi Bindport
Gordon Henderson wrote: On Mon, 29 Oct 2007, Abdul wrote: Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Really? What country?? What ISP? This isn't my experience in the UK .. (yet???) Gordon Many of them do that in North America. We've had lots of trouble with several providers: Time Warner, Rogers Cable, etc. What makes it more insidious in some cases is that they don't block ALL port 5060 traffic -- just INVITEs. Makes it incredibly difficult to debug when REGISTER messages come through just fine.. and options... and SUBSCRIBE/NOTIFY. But when you send an INVITE, it vanishes into the aether. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Virtual Appliances
Hi All, Does anyone know of a good virtual appliance for Asterisk under VMware? I am very interested in the JEOS concept for reducing the attack surface of a machine, so I think an appliance might be a good way to do this. BTW, I'll be using this with direct SIP Trunking and Snom 370/360 IP phones, so no hardware card is necessary. Thanks in advance! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Virtual Appliances
Zaheer Master wrote: Hi All, Does anyone know of a good virtual appliance for Asterisk under VMware? I am very interested in the JEOS concept for reducing the attack surface of a machine, so I think an appliance might be a good way to do this. BTW, I'll be using this with direct SIP Trunking and Snom 370/360 IP phones, so no hardware card is necessary. Thanks in advance! Regards, Zaheer K. Master President, Adamant Security Inc. http://www.vmware.com/appliances/directory/576 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to run ztcfg manually?
Tzafrir Cohen wrote: On Fri, Oct 26, 2007 at 04:52:07PM -0800, Mojo with Horan Company, LLC wrote: I don't have T1 but it seems that the first time I run ztcfg (or in fact, the zaptel startup script runs it for me) it fails. What distribution is it? RHEL4 / CentOS4 has an early udev version that seems to react quite slowly. Precisely, Centos4.4 -- A delay loop _would_ clean things up a bit, agreed. Thanks! For that reason that zaptel init.d script includes a delay loop. In earlier versions it had waited up to 10 seconds for /dev/zap/ctl to appear. In current versions it waits up to 20 seconds, and that number is configurable through /etc/sysconfig/zaptel (or /etc/default/zaptel on Debian). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Leg Control on Asterisk Callback
Why dont you make 2 separate Originate actions, one for each call leg. Then call Bridge manager Action whenever you want. Moy On 10/29/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 29, 2007 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP phone recommendation (used to be: no subject)
My apologies to the list for not having entered a subject line in the email. Thanks On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uniden UIP200 phones
Lyle Giese wrote: Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Do you mean you're using SetVar(Alert-Info: ...) instead of SIPAddHeader(Alert-Info: ...) ? Thanks, Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck Voicemails?
We used to have this problem with 1.2, too. I think it was some timing thing that resulted from the caller hanging up at just the right (or should I say, wrong) moment, like after the min-message-len timer. I won't tell you what we did to fix it, because you don't want to hear about upgrading to 1.4! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 29 Oct 2007, Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP multi Bindport
On Mon, 29 Oct 2007, Abdul wrote: Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Really? What country?? What ISP? This isn't my experience in the UK .. (yet???) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFC/R2 on AsteriskNOw
MFC/R2 on AsteriskNOw!! How? Please!!! Thanks!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stuck Voicemails?
This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fetch call
Our features.conf let us set *8 to pick up a ringing line elsewere in the system. I believe it can be extended to *8x, to pick up a specific group. moj Nuno Fernandes wrote: Hi, I have asterisk installed. When a connection comes from the outside one of our phones rings for about 45 seconds. Is it possible to another phone fetch the call while it's ringing on the first phone? I don't want to use ringgroups because the second phone would be ringing also. Thanks Nuno Fernandes ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fetch call
Nuno Fernandes wrote: Hi, I have asterisk installed. When a connection comes from the outside one of our phones rings for about 45 seconds. Is it possible to another phone fetch the call while it's ringing on the first phone? I don't want to use ringgroups because the second phone would be ringing also. Thanks Nuno Fernandes Check out the Pickup dial plan application. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck Voicemails?
Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6 asterisk? By stuck, I mean the phones show a I would suggest at least upgrading to the current 1.2.x series, currently 1.2.24 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP phone recommendation (used to be: no subject)
Well, just general office use. They are a real-state construction company, so the phones will get some heavy use since most of the phones are going to sales associates. Now, one of the things we are most interested in are: 1) Asterisk compatibility 2) Mass provisioning 3) Remote management 4) Excellent audio quality (I know there are many factors involved, but would like to rule out the phone set itself) 5) Robustness 6) Vendor reputation and warranties We have used Linksys 941s in the past and think they're pretty good. However, we've only used them in 3-5 phones office environments. We've also used the Polycoms IP 501 and 650s. They seem good, but sometimes the users complain about the audio being a bit weird in the sense that, probably, the silence detection may give the user a feeling that the line dropped. Then again, we've only used these once (one client installation for each), so for practical purposes, we don't really have any larger quantity real-life experience. Thanks On Oct 29, 2007, at 2:18 PM, Eric Chamberlain wrote: What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 29, 2007 10:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck Voicemails?
We have that problem here with Asterisk 1.2.9.1. There is a fix in later versions of the 1.2 branch, but I couldn't tell you which one. You can just delete the .txt file from the user's voicemail folder and it should clear the MWI on the phone. On 10/29/07, Matt [EMAIL PROTECTED] wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Stay away from Cisco they just don't work for the price, if it would be in the price range of a Grandstream phone I would tell you go for it, but at the current price its just not worth it. Aastra, Polycom or linksys all work for me. Never tried Snom before. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 on AsteriskNOw
just install chan_unicall.so On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: MFC/R2 on AsteriskNOw!! How? Please!!! Thanks!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck Voicemails?
Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6 asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? I don't know why it happens but we have run a very early 1.2 svn before jumping to 1.2.17 (fixed several issues) and incrementally through to 1.2.24 and we have not seen this issue. Any related errors in asterisk logs or system logs? Has fsck been run recently on the relevant filesystem? Is your mail stored on a local filesystem? Would upgrading to 1.2.24 be an acceptable upgrade? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Leg Control on Asterisk Callback
Hi, On Mon, 2007-10-29 at 10:29 -0700, Douglas Garstang wrote: I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? You can use dial macro here like exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],M(a_leg)) and [macro_a_leg] exten = s,1,Playback(tt-monkeys) you can run most of asterisk dialplan commands in macro. as soon as your macro finished your call will be connected to Leg B you can read more at http://www.voip-info.org/wiki-Asterisk+cmd+Dial#Dialmacros and http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro Nasir Iqbal ICT Innovations http://www.ictinnovations.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck Voicemails?
Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? IIRC there was a bug in early 1.2 releases that caused this problem. Upgrading to the latest 1.2.x fixed the issue for me. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia E65 SIP/2.0 407 Proxy Authentication Required Problem
Abdul wrote: Hi friends, We have are getting SIP/2.0 407 Proxy Authentication Required on Invite pakcet once Nokia E65 trying to dial number. But it can recive well from other caller. We tried to disable secrete and it worked fine. But we have lot of users and disabling secrete is risky. Interesting thing is Nokia N95, N80 is working well with the secrete the problem is only with Nokia E65. I will be appreciate if some one can help us to solve this issue. Check this setup instruction: http://wiki.diamondcard.us/NokiaE70 It may be helpful for your setup too. Dmitry __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
No: register = abc:[EMAIL PROTECTED] [peer] host=zzz Its possible to make mistakes and typos you know. Maybe you can post your config file and we can help you. On 10/26/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Pablo; How the IP address will be wrong, and asterisk able to do registeration on the destination? If the IP address wrong, so I will not be able to register on that IP address. Regards Bilal Hi List; Ip address to destination? Unable to create channel of type SIP (cause 3 - No route to destination) i think you have the wrong ip information I established an SIP IP Trunk between Asterisk and another softswitch (asterisk registered on the softswitch successfully) and I saw this on the softswitch. From firefly softphone, I was need to do a call to be via this softswitch (ofcourse, the softphone will send for asterisk and asterisk should route to the softswitch based on the extensions.conf configurations. But, always I receive this message (and the call does not even reach to the softswitch, it is not sended from Asterisk to the softswitch): Executing [EMAIL PROTECTED]:1] Dial(SIP/EgyptOeratorSIP-09f9bed0, SIP/[EMAIL PROTECTED]) is new stack Unable to create channel of type SIP (cause 3 - No route to destination) Everyone is busy/congested at this time (1:0/0/1) Anyone faced that? Is it related to a paramater that control number of allowed channels per IP trunk? Maybe I have such parameters is 0 ? I do not know even if there is such parameter. At the softswitch, I do not see even any attempt (nothing related to the dialed number), so why Asterisk does not send the called number to the softswitch and why asterisk assume there is not available channel? The softphone codec is g729a and the softswitch support such codec. Also, if it is a codec matter, then call should be send to the softswitch, and the softswitch will gives an error related to the codec missmatch. Any help? Regards Bilal Ghayad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Virtual Appliances
I suppose the VMware image of AsteriskNow is a good place to start? I just found this and I think it answers my question :) Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 weirdness and rejected calls: Invalid BYTE
All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EE where 'EE' is the exchange number and '' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public providers and 'exchanges' as leaf-nodes. This allows centralised call rouing with only a single entry in iax.conf at the edges and a big, pattern based, routing table in the middle. As long as everyone accepts the same codecs then the central box hands off the calls and the endpoints talk directly. A little over a week ago I upgraded my home box to Asterisk 1.4.12 - I have been using 1.4.xx here for some time but the rest of the boxes lag and are typically 1.2.16. Shortly after upgrading my home box (gate.tubby.org) - known as [tubby] in the config files - calls in from SIPgate to DDI numbers at home stopped working. Also calls from remote phones on another exchange to me stopped working - both resulted in the re-order tone (fast busy) and if called from a GSM phone then the mobile would display Call not allowed - for this the call path was: T-mobile --[GSM/Q.931/SS7]-- Magrethea Telecom --[SIP]-- asterisk.thorcom.net --[IAX2]-- gate.tubby.org If I ran debug on the central box (asterisk.thorcom.net) I could clearly see the call coming in and being placed on gate.tubby.org but it was being rejected with the message: [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:770 iax_error_output: Expecting causecode to be single byte but was 2 [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:7450 socket_process: Call rejected by 193.82.116.194: No supported codec found Now, over at gate.tubby.org a 'tcpdump' clearly showed the exchange of IAX packets, but enabling debug on IAX showed nothing!? I upgraded both gate.tubby.org and asterisk.thorcom.net to Asterisk 1.4.13 and tried again -- same results -- now confused I set about further testing to see what was going on and it just magically mended itself and started working... Here's a trace I had running over at asterisk.thorcom.net of a call failing: -- Unregistered IAX2 'vikki' (UNAUTHENTICATED) ;; this is a friend's Zoiper soft phone registering at the same time Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 4ms SCall: 2 DCall: 10100 [193.82.116.194:4569] USERNAME: vikki DATE TIME : 2007-10-29 19:46:36 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 4ms SCall: 10100 DCall: 2 [193.82.116.194:4569] -- Executing [EMAIL PROTECTED]:1] SIPDtmfMode(SIP/213.166.5.134-086112f8, inband) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/213.166.5.134-086112f8, 01905888007 ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/213.166.5.134-086112f8, IAX2/tubby/888007) in new stack -- Called tubby/888007 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00017ms SCall: 1 DCall: 0 [193.82.116.194:4569] VERSION : 2 CALLED NUMBER : 888007 CODEC_PREFS : (alaw|ulaw) CALLING NUMBER : 07939465009 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: 07939465009 LANGUAGE: en FORMAT : 8 CAPABILITY : 57356 ADSICPE : 2 DATE TIME : 2007-10-29 19:47:16 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00017ms SCall: 10101 DCall: 1 [193.82.116.194:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 5ms SCall: 10101 DCall: 1 [193.82.116.194:4569] CAUSE : No supported codec found CAUSE CODE : Invalid BYTE [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:770 iax_error_output: Expecting causecode to be single byte but was 2 [Oct 29 19:47:16] WARNING[16974]: chan_iax2.c:7450 socket_process: Call rejected by 193.82.116.194: No supported codec found Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 5ms SCall: 1 DCall: 10101 [193.82.116.194:4569] -- Hungup 'IAX2/tubby-1' [Oct 29 19:47:16] NOTICE[17128]: cdr.c:434 ast_cdr_free: CDR on channel 'IAX2/tubby-1' not posted == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/213.166.5.134-086112f8, ) in new stack == Spawn extension (sip-default, 01905888007, 4) exited non-zero on 'SIP/213.166.5.134-086112f8' So... was the problem no supported codec or invalid BYTE ? And a short while later, without even stopping or re-starting anything it started working again: -- Executing [EMAIL PROTECTED]:1] SIPDtmfMode(SIP/213.166.5.134-0860a750, inband)
Re: [asterisk-users] Asterisk: No Longer Answering Calls
On Mon, Oct 29, 2007 at 03:44:13PM +, Jeng Yu wrote: Hi Friends! I need help! I'm still Asterisk rookie, so please forgive me. My Asterisk is no longer answering incoming call on the phone line. I call the phone and it rings but asterisk is not picking it up. The phone line is attached to port 4 (FXO) on my digium TDM411P card. I am running Asterisk 1.4.11 with zaptel-1.4.5.1 and libpri-1.4.1 on Fedora Core 5, Linux Kernel 2.6.15-1.2054_FC5smp when I look in the asterisk messages log file I see this interesting messages: - [Oct 29 12:03:09] WARNING[2073] config.c: Unterminated comment detected beginning on line 28 Quoting doc/confguration.txt: The ;-- is a marker for a multi-line comment. Everything after that marker will be treated as a comment until the end-marker --; is found. Parsing begins directly after the end-marker. ;This is a comment label = value ;-- This is a comment --; [Oct 29 12:03:09] WARNING[2073] chan_zap.c: Unable to specify channel 4: No such device or address [Oct 29 12:03:09] ERROR[2073] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 [Oct 29 12:03:09] ERROR[2073] chan_zap.c: Unable to register channel '4' Please provide the output of: cat /proc/zaptel/* -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
doesn't look legit to me. It's got CE/FCC emblems, but no ID #'s ?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Sexton Sent: Monday, October 29, 2007 5:35 PM To: Asterisk Users Mailing List Subject: [asterisk-users] Mystery phone! Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mystery phone!
Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Mysql error
Hi: Iam using Fedora core 5 . Thanks in advance; Date: Mon, 29 Oct 2007 10:23:28 +0530 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime Mysql error On 10/27/07, wassim darwish wrote: Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command realtime mysql status on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds. Can any body help with this; Hi what is the version of asterisk and mysql what distro you are using ram _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
I've had experience with Linksys and Polycom. Either one is easy enough to provision. Took me a while to understand how to provision Polycom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 30 October 2007 3:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13
On Mon, Oct 29, 2007 at 10:19:49AM +, Steve Davies wrote: snom phones have been using ports in the 2000+ range since the dawn of asterisk without any problems, so I suspect that this will be an Asterisk configuration error, or a change to the asterisk SIP stack that is causing problems. I suspect the latter. As it turns out, setting nat=yes in the sip.conf entry causes Asterisk to use the correct port. The phone is not behind a NAT, so nat=no should work, but it doesn't. This is a very supportive group; thanks for the prompt assistance. Can you also check that the snom has a suitably recent firmware version. It may be a bug in something the phone is sending. Yes, it's a recent version loaded onto a new phone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
If you go to linksys's website and click on partners then apply for partnership you will be able to get access to the documents programs you need John [EMAIL PROTECTED] wrote: Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
Or you can download them at http://spc.pifiu.com and not have to go through that bullshit. On 10/29/07, John Mason Jr [EMAIL PROTECTED] wrote: If you go to linksys's website and click on partners then apply for partnership you will be able to get access to the documents programs you need John [EMAIL PROTECTED] wrote: Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
Hmm the shape looks like an Aastra but the buttons down the side look like PlayStation buttons to me. Maybe it's a Sony Cisco joint effort. Joel. On Mon, 2007-10-29 at 16:35 -0500, Kyle Sexton wrote: Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP phone recommendation (used to be: no subject)
On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote: Well, just general office use. They are a real-state construction company, so the phones will get some heavy use since most of the phones are going to sales associates. Now, one of the things we are most interested in are: 1) Asterisk compatibility 2) Mass provisioning 3) Remote management 4) Excellent audio quality (I know there are many factors involved, but would like to rule out the phone set itself) 5) Robustness 6) Vendor reputation and warranties We have used Linksys 941s in the past and think they're pretty good. However, we've only used them in 3-5 phones office environments. We've also used the Polycoms IP 501 and 650s. They seem good, but sometimes the users complain about the audio being a bit weird in the sense that, probably, the silence detection may give the user a feeling that the line dropped. Then again, we've only used these once (one client installation for each), so for practical purposes, we don't really have any larger quantity real-life experience. For my money it's Polycom every time. It's great hardware. Meets all your requirements. I thought that silence supression was specifically disallowed with Asterisk? Something about timing requirements not being met. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime context
Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include = parkedcalls switch = Realtime/@ [fromiax] switch = Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch = Realtime/@ but I have about 50 asterisk that read one database, now if I want to change/add a context I have to change 50 extensions.conf file :( The easy answer to your question is, no, you can not put a context in the database without a corresponding context in the static extensions.conf file. You need: [newcontext] switch = Realtime/@ or it will not work. The hard answer is, yes, there is a catch all patch out there is the ether, do a google search for asterisk alf scherer and you can catch up on the progress with the patch, try it, it may work for you. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
Smith, Rick wrote: doesn't look legit to me. It's got CE/FCC emblems, but no ID #'s ?! If that is a mark of legitimacy, then most equipment must be fake. :-) Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDI setup help
Could anybody help? Can you show a CLI session? The error you get is not familiar. Otherwise your configs look ok, did you make the keys priv or dundi? There was an error in the howto, the example was to make the keys named priv but in dundi.conf the keys were named dundi, double check that as well. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 from RPM
That's really a question for [EMAIL PROTECTED] The short and generally not very helpful answer is that there are a lot of poorly packaged software releases out there that don't play well with cross-development environments. -Philip Douglas Garstang wrote: I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk in SIP trunking mode with a Coppercom switch
Has anyone had any experience in getting Asterisk to interoperate with a Coppercom switch using SIP, either as subscriber lines or else as a trunked configuration? And if so, do you have any configs you could share (for both ends)? Thanks, -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 from RPM
I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 from RPM
Since I'm executing a 'make rpm' from within the Asterisk 1.4.13 distribution source, I'd say it's an Asterisk question. - Original Message From: Philip Prindeville [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 29, 2007 6:24:06 PM Subject: Re: [asterisk-users] Asterisk 1.4 from RPM That's really a question for [EMAIL PROTECTED] The short and generally not very helpful answer is that there are a lot of poorly packaged software releases out there that don't play well with cross-development environments. -Philip Douglas Garstang wrote: I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64.. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uniden UIP200 phones
Mojo with Horan Company, LLC wrote: Lyle Giese wrote: Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Do you mean you're using SetVar(Alert-Info: ...) instead of SIPAddHeader(Alert-Info: ...) ? Thanks, Moj I WAS using SetVar with * v1.0.x. For version 1.4.x, I had to ask what the new syntax was for the same functionality. Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
I don't know that I would want to download an unauthorized copy of a program to run on my computer without means to verify it's authenticity. And even if the programs and docs are valid, why not sign up and get them from the source, might even be beneficial. John [EMAIL PROTECTED] wrote: Or you can download them at http://spc.pifiu.com and not have to go through that bullshit. On 10/29/07, John Mason Jr [EMAIL PROTECTED] wrote: If you go to linksys's website and click on partners then apply for partnership you will be able to get access to the documents programs you need John [EMAIL PROTECTED] wrote: Take a look at http://spc.pifiu.com there they have the spc.exe ( Linux variant) which will generate the sample XML file for your firmware version. There is also in PDF format the admin guides that explain all the parameters. On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
On Oct 29, 2007, at 5:35 PM, Kyle Sexton wrote: Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg I don't know who makes the above phone, but physically, it looks nearly identical to the SBC 125 or SBC 225 http:// www.sbcphonestore.com/SBC-Corded-Telephones/1-Line-Multifunction- Caller-ID-Speakerphone-SBC-125-ii_2.html I have no idea if SBC makes their phone themselves or contract it out to someone else. But going just off look, I'd think the SBC phone and your mystery phone clearly have some part of the manufacturing process in common, because it is definitely using the same shell. I have access to a few of the SBC 120 (also the same case, but lacks the little side panel for speed dial info), so if you really need to know more, I can look for FCC numbers or other info to try to determine who the ultimate manufacturer is for the SBC phone. -chris www.mythtech.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
The web site is Russian (Serbian I think). Company is Hybird Systems (Hibridni System AD). Best I can tell which probably does not help much except to say it is a legit company that has been around a long time making computer stuff since the 60's. On 10/30/07, Kyle Sexton [EMAIL PROTECTED] wrote: Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime context
Maybe I'm not following the problem here ... couldn't he just rework his extensions in a way that uses macros so he doesn't have to change 50 things? On 10/30/07, JR Richardson [EMAIL PROTECTED] wrote: Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include = parkedcalls switch = Realtime/@ [fromiax] switch = Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch = Realtime/@ but I have about 50 asterisk that read one database, now if I want to change/add a context I have to change 50 extensions.conf file :( The easy answer to your question is, no, you can not put a context in the database without a corresponding context in the static extensions.conf file. You need: [newcontext] switch = Realtime/@ or it will not work. The hard answer is, yes, there is a catch all patch out there is the ether, do a google search for asterisk alf scherer and you can catch up on the progress with the patch, try it, it may work for you. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Leg Control on Asterisk Callback
Read all the options of the Dial() function. There are options you can mess with to play something while the call is ringing (music on hold feature if I recall). Check out all the Dial options. On 10/29/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
