Re: [asterisk-users] need * consultant in houston area
Could you all please take this COMMERCIAL discussion to the -biz list? Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Newbie MeetMe: How to control max users in conference?
> Yes, use your first solution, but precede it with a call to the Read()
> application for the user to enter their conference number. This will
put
> it into a channel variable, e.g. ${CONF}, which you can then put in
place
> of the hard coded number.
Thanks Tony for your advice.
Below is a working version of a Meetme extension which a) check the
conference room number and b) restrict the max size of the attendees of
each conference room.
The good thing is that we only need to program one extension for all
conference room instead of one for each.
The down side is that I have to hard code the conference room numbers
and the max size (which is not good from a good programmer's point of
view).
exten => 8101,1,Answer()
exten => 8101,n(L8101A),Playback(enter-conf-call-number)
exten => 8101,n,Read(ConfNumber,,3)
exten => 8101,n,GotoIf($[${ConfNumber} = 101]?L8101D:L8101B)
exten => 8101,n(L8101B),GotoIf($[${ConfNumber} = 102]?L8101D:L8101C)
exten => 8101,n(L8101C),Playback(conf-invalid)
exten => 8101,n,Wait(1.5)
exten => 8101,n,Goto(L8101A)
exten => 8101,n(L8101D),Playback(conf-thereare)
exten => 8101,n,MeetMeCount(${ConfNumber},ConfCount)
exten => 8101,n,SayNumber(${ConfCount})
exten => 8101,n,Playback(conf-peopleinconf)
exten => 8101,n,GotoIf($[${ConfCount} < 10]?L8101E:L8101F)
exten => 8101,n(L8101E),MeetMe(${ConfNumber},ciMps)
exten => 8101,n,Playback(vm-goodbye)
exten => 8101,n,Hangup()
exten => 8101,n(L8101F),Playback(conf-full)
exten => 8101,n,Playback(vm-goodbye)
exten => 8101,n,Hangup()
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[asterisk-users] Little help with Conference
These is my scenario. Asterisk 1.4.16 Zaptel1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make reverse for these??? Thanks Ruben ___ -- 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 * consultant in houston area
Is it mandatory that the consultant be in the Houston area, can we work from a remote location such as Omaha, NE ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone Sent: Monday, March 10, 2008 8:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] need * consultant in houston area pls kindly respond to this email thx ! _ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008 ___ -- 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] need * consultant in houston area
What kind of help are you needing? On Mon, Mar 10, 2008 at 8:40 PM, A_ Navone <[EMAIL PROTECTED]> wrote: > > pls kindly respond to this email > thx ! > > _ > Connect and share in new ways with Windows Live. > http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008 > ___ > -- 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 > -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.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] Microsoft Office Communications Server
Kristian Kielhofner wrote: > On Mon, Mar 10, 2008 at 6:38 PM, <[EMAIL PROTECTED]> wrote: >> What is the logic of them using SIP over TCP? Is this a broad industry >> trend? Or just the latest attempt to get around SIP/NAT issues? >> >> Michael Graves >> mgraves mstvp.com >> o(713) 861-4005 >> c(713) 201-1262 >> sip:[EMAIL PROTECTED] >> skype mjgraves >> FWD 54245 >> > > I would imagine it's because they plan on doing all kinds of "neat" > stuff with SIP including video, TXT, Windows Updates, who knows... > SIP over UDP has some pretty serious packet fragmentation issues. If > you end up with a large enough SDP or something that causes a SIP > packet to grow larger than the smallest MTU in the path between your > two endpoints it doesn't work (no fragmentation support with SIP over > UDP). SIP over TCP does not have this problem. > > Also, you need SIP TCP support for TLS... > Well... I have been a MS windows desktop user for a while as many other people have. It mostly works except at times one needs to maintain/repair what one bought. I have switched :) Imagine, repairing an engine of your brand new car you just bought? Imagine "restarting" your TV because it just froze? What if your shoes have "just" changed colour to "blue screen"? It will just not "pass", will it? ... You will DEMAND a service for your car/TV,shoes or you may return it or whatever. So.. Imagine how much your business will be affected with a phone SYSTEM based on a such operating system, one which can not even meet basic desktop user requirements let alone crucial every day in/out business communications tool like a phone system. At the end, if you do not answer a call some else will!!! Senad Jordanovic www.bicomsystems.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] need * consultant in houston area
pls kindly respond to this email thx ! _ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008 ___ -- 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] replace astdb with a cluster-capable sql database engine
Hi! > So putting a translation layer so that ast_db_* API calls either go the > normal route or translate to func_odbc (or another path) would improve > functionality because both old and new apps would be able to seamlessly > take advantage of the new database backend or keep using DB1 (the * > admin would decide). This is not exactly what you are aiming at, and probably outdated, but still close enough to be of interest, I assume: http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc Cheers, Philipp ___ -- 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] Microsoft Office Communications Server
Hi! > What is the logic of them using SIP over TCP? Is this a broad industry > trend? Or just the latest attempt to get around SIP/NAT issues? I remember a quote of Henning Schulzrinne where he states that having designed SIP with UDP in mind was the biggest mistake he (and Mark Handle?) were to be found guilty of. I am not sure if this is what's driving Microsoft's decisions, my guess is that this is/was mostly driven by security reasons (and the new focus of Microsoft on security aspects). Cheers, Philipp * Taken from http://www.faqs.org/rfcs/rfc4168.html: 3.1. Advantages over UDP All the advantages that SCTP has over UDP regarding SIP transport are also shared by TCP. Below, there is a list of the general advantages that a connection-oriented transport protocol such as TCP or SCTP has over a connection-less transport protocol such as UDP. Fast Retransmit: SCTP can quickly determine the loss of a packet, because of its usage of SACK and a mechanism that sends SACK messages faster than normal when losses are detected. The result is that losses of SIP messages can be detected much faster than when SIP is run over UDP (detection will take at least 500 ms, if not more). Note that TCP SACK exists as well, and TCP also has a fast retransmit option. Over an existing connection, this results in faster call setup times under conditions of packet loss, which is very desirable. This is probably the most significant advantage of SCTP for SIP transport. Congestion Control: SCTP maintains congestion control over the entire association. For SIP, this means that the aggregate rate of messages between two entities can be controlled. When SIP is run over TCP, the same advantages are afforded. However, when run over UDP, SIP provides less effective congestion control. This is because congestion state (measured in terms of the UDP retransmit interval) is computed on a transaction-by-transaction basis, rather than across all transactions. Thus, congestion control performance is similar to opening N parallel TCP connections, as opposed to sending N messages over one TCP connection. Transport-Layer Fragmentation: SCTP and TCP provide transport-layer fragmentation. If a SIP message is larger than the MTU size, it is fragmented at the transport layer. When UDP is used, fragmentation occurs at the IP layer. IP fragmentation increases the likelihood of having packet losses and makes NAT and firewall traversal difficult, if not impossible. This feature will become important if the size of SIP messages grows dramatically. * Quote from http://tools.ietf.org/html/draft-jennings-sip-dtls-01: There has been considerable discussion of why SIP needs DTLS when we have TLS. This is the wrong question. The right question is why SIP has UDP and TCP (not to mention SCTP). There are two reasons for believing that UDP is likely to be an important protocol in SIP for the foreseeable future. o In theory, there is no problem building systems that terminate a million TCP connections on a single host. In practice, the common operating systems used for building SIP aggregation devices make this impossible. To date, no one has demonstrated terminating over 100k SIP TCP connections to a single host. Doing that many connections with UDP has not been difficult. o If we want to talk about "running code" for SIP, it's UDP. Unless UDP is deprecated for SIP, it is important to provide a reasonable level of security for it. ___ -- 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] Intermittent DTMF Problems
I have seen too high of audio levels cause echo. It can also distort the audio. I imagine either of which I imagine the system can detect as a doubled digit. When I experienced this on some lines in Glufport, MS, random digits were doubled. He's tried everything else. Brent Davidson wrote: > Would TXgain really affect DTMF detection all that much on an incoming > call? I can see how RXgain might cause some problems if it was too high > or too low, but I adjusted both of these settings according to the echo > cancellation guide using the Type 102 Milliwatt test lines. My rxgain > is already -2.8 and if I drop my txgain below +4 callers complain that > they can't hear the users on the Sip phones inside the offices. > > Thanks, > Brent > > Eric Wieling wrote: >> Lower the rxgain and txgain on your Zap channels. >> >> bilal ghayyad wrote: >> >>> Hi Brent; >>> >>> I have been suffering from this problem since about 2 >>> monthes and until now still did not resolved 100%. >>> >>> First of all, I need to tell u that mostly u have a >>> problem that the first digit is duplicated, for >>> example: if ur customer entered 108 then it will be >>> recognized 110 (the 1 duplicated, and then it takes >>> the 0, and it does not continue to take the 8 as it >>> completes the 3 digits ... this is just an example). >>> >>> Your problem is in the duplication for the digit and >>> specifically the first digit usually will be >>> duplicated. >>> >>> If u found a solution let me know. >>> >>> Regards >>> Bilal >>> >>> >>> --- >>> >>> I've recently installed Asterisk-based servers at >>> several of our branch >>> >>> offices. Each server has 2 X100P cards to handle 2 >>> incoming voice lines. I was having a lot of trouble with Echo until >>> I got OSLEC running on all of the servers, but now we have a new >>> problem. Incoming >>> >>> callers are not always able to dial extensions. I >>> would say probably 95% of the calls go through correctly, but that other >>> 5% always get dumped to the operator queue. I have relaxdtmf=yes in >>> my zapata.conf for both channels, but it doesn't always help. One >>> particular customer >>> >>> has trouble dialing an extension about 1 call in 5. I'm wondering if >>> it's just because they call us more than any of our >>> other customers or if there is some peculiarity with their phone >>> system. Anybody have any >>> >>> ideas what to try next? >>> Thanks, >>> Brent Davidson >>> >>> >>> >>> >>> >>> >>> >>> Looking for last minute shopping deals? Find them fast with Yahoo! >>> Search. >>> http://tools.search.yahoo.com/newsearch/category.php?category=shopping >>> >>> ___ >>> -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Shared Extension
Raj,
I would say you understand exactly. It is kind of a SLA, but not.
SLA does great with a inbound trunk line and multiple extensions, but even in
SLA, if one extension is busy, the others ring.
There is no way to tell asterisk that if it gets a busy on one of the channels,
that the extension is busy, period.
The terminology to say that multiple extensions appear as a single extension is
not correct either. To say that you would have to define an extension in the
system and that each of these extension numbers is pooled in a Local type dial
command to the single extension. So because that terminology is not adequate,
I am using one extension to multiple channels.
I am trying to create a single extension to multiple channels (lines) {exten =>
5000,1,Dial(SIP\1234&SIP\phone&Local\12225551212)} but respecting busy on any
channel is busy on the extension. Almost the reverse of SLA, but with all the
behavior of a single extension to a single channel {exten =>
5000,1,Dial(SIP\1234)}
Thanks for working with me to clarify.
Tony Plack
> I don't quite understand the use case, but it sounds like you may
> be trying to do shared line appearances
> (http://asterisk.org/node/48342). You seem to be alluding that you
> want multiple extensions to share the state of a single extension.
> If that is the case, then SLA isn't quite that. Also, Asterisk SLA
> doesn't support a notion of call appearance where a single
> extension can receive multiple calls.
>
> --
> Raj
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Re: [asterisk-users] FaxBack Service with Asterisk
My back is far too hairy - I imagine all the hairs would just clog up the fax machine. PaulH On Mon, 2008-03-10 at 15:30 +0200, Dovid B wrote: > Hi, > Has anyone ever used asterisk for a faxback service ? > > Thanks. > > Dovid > ___ > -- 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] Microsoft Office Communications Server
On Mon, Mar 10, 2008 at 6:38 PM, <[EMAIL PROTECTED]> wrote: > What is the logic of them using SIP over TCP? Is this a broad industry > trend? Or just the latest attempt to get around SIP/NAT issues? > > Michael Graves > mgraves mstvp.com > o(713) 861-4005 > c(713) 201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > FWD 54245 > I would imagine it's because they plan on doing all kinds of "neat" stuff with SIP including video, TXT, Windows Updates, who knows... SIP over UDP has some pretty serious packet fragmentation issues. If you end up with a large enough SDP or something that causes a SIP packet to grow larger than the smallest MTU in the path between your two endpoints it doesn't work (no fragmentation support with SIP over UDP). SIP over TCP does not have this problem. Also, you need SIP TCP support for TLS... -- Kristian Kielhofner ___ -- 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] Microsoft Office Communications Server
On 15:38, Mon 10 Mar 08, [EMAIL PROTECTED] wrote: > What is the logic of them using SIP over TCP? Is this a broad industry > trend? Or just the latest attempt to get around SIP/NAT issues? Their setup implements some 'non standard extensions' on the SIP standard and I think it was easier to do it in TCP. (probably because they bought it from someone else, and that someone did it it TCP) Of course, because I'm not a MS developer that's only guessing. > > Michael Graves > mgraves mstvp.com > o(713) 861-4005 > c(713) 201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > FWD 54245 > > > > Original Message > > Subject: Re: [asterisk-users] Microsoft Office Communications Server > > From: "Kristian Kielhofner" <[EMAIL PROTECTED]> > > Date: Mon, March 10, 2008 5:18 pm > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > > > -BEGIN PGP SIGNED MESSAGE- > > > Hash: SHA1 > > > > > > Has anyone done any integration with this? > > > > > > All I know so far is that it appears to use some non standard form of > > > SIP. > > > > > > Any pointers? > > > > > > - -- > > > Kind Regards, > > > > > > Matt Riddell > > > Director > > Matt, > > I believe OCS only supports SIP over TCP. You'll either need to use > > Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP > > proxy. > > -- > > Kristian Kielhofner > > ___ > > -- 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 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called 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] Shared Extension
I don't quite understand the use case, but it sounds like you may be trying to do shared line appearances (http://asterisk.org/node/48342). You seem to be alluding that you want multiple extensions to share the state of a single extension. If that is the case, then SLA isn't quite that. Also, Asterisk SLA doesn't support a notion of call appearance where a single extension can receive multiple calls. -- Raj On Mon, Mar 10, 2008 at 11:00 AM, Tony Plack <[EMAIL PROTECTED]> wrote: > I am working on a project that requires shared extension. Where shared line > looks at the status of a line/trunk, shared extension would look at a series > of channels as the same "extension". > > The users would like to add destination channels on the fly, to provide > roaming extensions, but maintaining fixed channels as well. > > If a call comes in on an extension, the system needs to honor the fact that > channel 1 is busy, therefore, the extension is busy. Keep in mind that the > channel could be anything including SIP outbound trunk channels (read cell > phone or hotel room). > > The Dial command does provide a nice multi-channel dialer, especially with > the "r" option, however, if one of the lines is busy, the system will keep > ringing the other lines until timeout or answer (read voice mail). > > So I am contemplating adding a feature to the dial command, that would make > any channel busy, cause the initial Dial to come back as busy. Kind of a > force the state flag. > > Before I brake into code, does anyone have any other ideas? > > This would also help with phones like Grandstream, where you have 4 accounts > to configure, and would like to have all 4 SIP accounts act as 1 extension. > > Tony Plack > > ___ > -- 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 > -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot 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] Microsoft Office Communications Server
I would rather stick needles in my eyes but that's just me. -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Sunday, March 09, 2008 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Microsoft Office Communications Server -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1KVoDQNt8rg0Kp4RAgSfAJ0aXGIkKi6kGAjZK8TtSV2mMj79qQCdHhAS 1jZ9sjtsTJ3O1R9J3giztw8= =Mlnt -END PGP 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
Re: [asterisk-users] display time on Cisco 79xx
Thank you to everyone for their help, my Cisco phones are now showing the right time. I really appreciate your time everyone, especial Mark. No virus found in this outgoing message. Checked by AVG. Version: 7.5.518 / Virus Database: 269.21.7/1323 - Release Date: 3/10/2008 11:07 AM ___ -- 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] Microsoft Office Communications Server
What is the logic of them using SIP over TCP? Is this a broad industry trend? Or just the latest attempt to get around SIP/NAT issues? Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 > Original Message > Subject: Re: [asterisk-users] Microsoft Office Communications Server > From: "Kristian Kielhofner" <[EMAIL PROTECTED]> > Date: Mon, March 10, 2008 5:18 pm > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > > -BEGIN PGP SIGNED MESSAGE- > > Hash: SHA1 > > > > Has anyone done any integration with this? > > > > All I know so far is that it appears to use some non standard form of SIP. > > > > Any pointers? > > > > - -- > > Kind Regards, > > > > Matt Riddell > > Director > Matt, > I believe OCS only supports SIP over TCP. You'll either need to use > Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP > proxy. > -- > Kristian Kielhofner > ___ > -- 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] About CID with DTMF in Asterisk
Hi,
I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the
data is arriving to the asterisk but asterisk isn't interpretating it:
its my full log:
1.
Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0
2.
[Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple
switch on 'Zap/4-1'
3.
[Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:1] Set("Zap/4-1", "__FROM_DID=s") in new stack
4.
[Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:2] Wait("Zap/4-1", "4") in new stack
5.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF end '9' received on
Zap/4-1, duration 0 ms
6.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF begin emulation of
'9' with duration 100 queued on Zap/4-1
7.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF end emulation of '9'
queued on Zap/4-1
8.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF end '3' received on
Zap/4-1, duration 0 ms
9.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF begin emulation of
'3' with duration 100 queued on Zap/4-1
10.
[Mar 10 16:26:03] DTMF[9274] channel.c: DTMF end '6' received on
Zap/4-1, duration 0 ms
11.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '2'
queued on Zap/4-1
12.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'6' with duration 100 queued on Zap/4-1
13.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '8' received on
Zap/4-1, duration 0 ms
14.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '6'
queued on Zap/4-1
15.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'8' with duration 100 queued on Zap/4-1
16.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '1' received on
Zap/4-1, duration 0 ms
17.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '8'
queued on Zap/4-1
18.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '6' received on
Zap/4-1, duration 0 ms
19.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'1' with duration 100 queued on Zap/4-1
20.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '5' received on
Zap/4-1, duration 0 ms
21.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '1'
queued on Zap/4-1
22.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'6' with duration 100 queued on Zap/4-1
23.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '3' received on
Zap/4-1, duration 0 ms
24.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '6'
queued on Zap/4-1
25.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'5' with duration 100 queued on Zap/4-1
26.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end '2' received on
Zap/4-1, duration 0 ms
27.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end emulation of '5'
queued on Zap/4-1
28.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF begin emulation of
'8' with duration 100 queued on Zap/4-1
29.
[Mar 10 16:26:04] DTMF[9274] channel.c: DTMF end 'C' received on
Zap/4-1, duration 0 ms
30.
[Mar 10 16:26:05] DTMF[9274] channel.c: DTMF end emulation of '8'
queued on Zap/4-1
31.
[Mar 10 16:26:05] DTMF[9274] channel.c: DTMF begin emulation of
'6' with duration 100 queued on Zap/4-1
32.
[Mar 10 16:26:05] DTMF[9274] channel.c: DTMF end emulation of '6'
queued on Zap/4-1
33.
[Mar 10 16:26:05] DTMF[9274] channel.c: DTMF begin emulation of
'C' with duration 100 queued on Zap/4-1
34.
[Mar 10 16:26:05] DTMF[9274] channel.c: DTMF end emulation of 'C'
queued on Zap/4-1
35.
[Mar 10 16:26:07] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:3] Set("Zap/4-1", "CALLERID(name)=") in new stack
36.
[Mar 10 16:26:07] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:4] NoOp("Zap/4-1", "CallerID is "" <>") in new stack
37.
[Mar 10 16:26:07] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:5] Set("Zap/4-1", "FAX_RX=disabled") in new stack
38.
[Mar 10 16:26:07] VERBOSE[9274] logger.c: -- Executing
[EMAIL PROTECTED]:6] Goto("Zap/4-1", "ivr-2|s|1") in new stack
39.
[Mar 10 16:26:07] VERBOSE[9274] logger.c: -- Goto (ivr-2,s,1)
Its my zapata.conf
[channels]
;;General options
relaxdtmf=yes
busydetect=yes
busycount=4
immediate=no
cidsignalling=dtmf
cidstart=polarity
sendcalleridafter=2
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
rxgain=6%
txgain=7%
callerid=asreceived
;;FXO Modules
Group=1
echocancel=yes
signalling=fxs_ks
context=from-trunk
channel=4
Thanks for your help.
--
José David Bravo Álvarez
ColombiaHosting E.U.
Av. 4 Norte #48n43 of.
Re: [asterisk-users] Intermittent DTMF Problems
Would TXgain really affect DTMF detection all that much on an incoming call? I can see how RXgain might cause some problems if it was too high or too low, but I adjusted both of these settings according to the echo cancellation guide using the Type 102 Milliwatt test lines. My rxgain is already -2.8 and if I drop my txgain below +4 callers complain that they can't hear the users on the Sip phones inside the offices. Thanks, Brent Eric Wieling wrote: Lower the rxgain and txgain on your Zap channels. bilal ghayyad wrote: Hi Brent; I have been suffering from this problem since about 2 monthes and until now still did not resolved 100%. First of all, I need to tell u that mostly u have a problem that the first digit is duplicated, for example: if ur customer entered 108 then it will be recognized 110 (the 1 duplicated, and then it takes the 0, and it does not continue to take the 8 as it completes the 3 digits ... this is just an example). Your problem is in the duplication for the digit and specifically the first digit usually will be duplicated. If u found a solution let me know. Regards Bilal --- I've recently installed Asterisk-based servers at several of our branch offices. Each server has 2 X100P cards to handle 2 incoming voice lines. I was having a lot of trouble with Echo until I got OSLEC running on all of the servers, but now we have a new problem. Incoming callers are not always able to dial extensions. I would say probably 95% of the calls go through correctly, but that other 5% always get dumped to the operator queue. I have relaxdtmf=yes in my zapata.conf for both channels, but it doesn't always help. One particular customer has trouble dialing an extension about 1 call in 5. I'm wondering if it's just because they call us more than any of our other customers or if there is some peculiarity with their phone system. Anybody have any ideas what to try next? Thanks, Brent Davidson Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Intermittent DTMF Problems
I was having the digit duplication early on, but turning the relaxdtmf option on and X Windows off solved the duplication problem. I have logging turned up extremely high and there are no digits detected on the calls that are unable to dial an extension. The way I have my dial plan set up dialing an extension that does not exist sends the user to an "Invalid Extension" message then returns them to the main menu. When a caller has issues with the DTMF the logs show that the WaitExten timed out with no digits dialed. Thanks, Brent bilal ghayyad wrote: > Hi Brent; > > I have been suffering from this problem since about 2 > monthes and until now still did not resolved 100%. > > First of all, I need to tell u that mostly u have a > problem that the first digit is duplicated, for > example: if ur customer entered 108 then it will be > recognized 110 (the 1 duplicated, and then it takes > the 0, and it does not continue to take the 8 as it > completes the 3 digits ... this is just an example). > > Your problem is in the duplication for the digit and > specifically the first digit usually will be > duplicated. > > If u found a solution let me know. > > Regards > Bilal > > > --- > > I've recently installed Asterisk-based servers at > several of our branch > > offices. Each server has 2 X100P cards to handle 2 > incoming voice > lines. I was having a lot of trouble with Echo until > I got OSLEC > running on all of the servers, but now we have a new > problem. Incoming > > callers are not always able to dial extensions. I > would say probably > 95% of the calls go through correctly, but that other > 5% always get > dumped to the operator queue. I have relaxdtmf=yes in > my zapata.conf > for both channels, but it doesn't always help. One > particular customer > > has trouble dialing an extension about 1 call in 5. > I'm wondering if > it's just because they call us more than any of our > other customers or > if there is some peculiarity with their phone system. > Anybody have any > > ideas what to try next? > > Thanks, > Brent Davidson > > > > > > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > > ___ -- 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] Microsoft Office Communications Server
On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Has anyone done any integration with this? > > All I know so far is that it appears to use some non standard form of SIP. > > Any pointers? > > - -- > Kind Regards, > > Matt Riddell > Director Matt, I believe OCS only supports SIP over TCP. You'll either need to use Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP proxy. -- Kristian Kielhofner ___ -- 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] Redirecting channels?
What interfaces you Dialogic box has ? 2008/3/10, harry <[EMAIL PROTECTED]>: > > Hello > > I am going to have a setup like this: > > One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the > other hand, I also have another box with VoiceGuide and Dialogic. As a > temporary migration-solution i would like to redirect some of the > ISDN30 channels from the Asterisk to the Dialogic-box. > > How would I do this? > > ___ > -- 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] Global Variables on Reload
or use astdb, pretty much simple 2008/3/10, Edwin Lam <[EMAIL PROTECTED]>: > > Rob Schall wrote: > > I'm running Asterisk 1.4.18 and having a problem with the > > clearglobalvars option. > > > > I have a NIGHT_SERVICE variable which I initially set equal to off. I > > then have an extension they can dial which will toggle that variable. My > > problem is when you enter the CLI and type "reload", it resets to "off" > > again. I've tried setting the clearglobalvars=no as well as just > > commenting out that line, but no luck so far. > > > > Any ideas? > > > we use MySQL db to store those global vars in our installation. i > guess you can use and db to do that. > > > > -- > Edwin Lam <[EMAIL PROTECTED]> > Systems Engineer, Office General, Inc. > Ph: +1 415 439 4988 Fax: +1 415 283 3370 > http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 > > > > ___ > -- 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 show channels - gives a growing list of dead channels
Asterisk SIP channels can hang for a variety of reasons such as network errors, signaling malfunction and software bugs. These are difficult to track down and sometimes the root cause is not even in your control. In order to provide a sort of "garbage collection" mechanism for such hung SIP channels, Asterisk 1.6 supports a mechanism called as SIP Session Timers. You may want to give this feature a shot. The instructions for configuring it are in sip.conf. -- Raj On Mon, Mar 10, 2008 at 5:13 PM, Keith Hardee <[EMAIL PROTECTED]> wrote: > I feel like I've seen that error before, but I did some quick testing > and was not able to produce the error. CLI level was greater than 206 > (many v's) > > callfromto hangup > Test 1polycom spectralink polycom > Test 2polycom spectralink spectralink > Test 3spectralink polycom polycom > Test 4spectralink polycom spectralink > Test 5 spectralink spectralink spectralink > > I only did one test of each above because I am not in office (had > someone doing tests while I watched CLI). I can test more when I get > back Thursday. > > Thanks for input. > > > > > On Sat, Mar 8, 2008 at 2:59 AM, Fons van der Beek > <[EMAIL PROTECTED]> wrote: > > Same problem over here > > > > I use KIRK-Telecom ip600v3 > > This only happens on calls between SIP en MiSDN, anyone any clue? > > > > As far as i can see these dead calls once in while occur when the > > remote party first hangs up (remote=MiSDN channel) > > > > Keith do you also have error messages in the CLI when you open asterisk > > by using asterisk > > -rvv ? (a lot of > v) > > > > -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71 > > > > 10.0.0.71 represents the IP number of internal phone > > > > Keith Hardee schreef: > > > > > > > I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 > > > Spectralink wireless IP phones. > > > > > > Most of the Spectralink phones have entries in 'sip show channels' > > > that do not go away. None of the other phones do this. > > > > > > Is there anyway to remove these entries without restarting Asterisk? > > > > > > Any ideas on what could be done to prevent this? > > > > > > Example output: > > > xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) > > > No (d) Rx: BYE > > > xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) > > > No Rx: BYE > > > xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) > > > No Rx: BYE > > > > > > Thanks, > > > Keith > > > > > > ___ > > > -- 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 > -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot 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] sip show channels - gives a growing list of dead channels
I feel like I've seen that error before, but I did some quick testing and was not able to produce the error. CLI level was greater than 206 (many v's) callfromto hangup Test 1polycom spectralink polycom Test 2polycom spectralink spectralink Test 3spectralink polycom polycom Test 4spectralink polycom spectralink Test 5 spectralink spectralink spectralink I only did one test of each above because I am not in office (had someone doing tests while I watched CLI). I can test more when I get back Thursday. Thanks for input. On Sat, Mar 8, 2008 at 2:59 AM, Fons van der Beek <[EMAIL PROTECTED]> wrote: > Same problem over here > > I use KIRK-Telecom ip600v3 > This only happens on calls between SIP en MiSDN, anyone any clue? > > As far as i can see these dead calls once in while occur when the > remote party first hangs up (remote=MiSDN channel) > > Keith do you also have error messages in the CLI when you open asterisk > by using asterisk > -rvv ? (a lot of v) > > -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71 > > 10.0.0.71 represents the IP number of internal phone > > Keith Hardee schreef: > > > > I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 > > Spectralink wireless IP phones. > > > > Most of the Spectralink phones have entries in 'sip show channels' > > that do not go away. None of the other phones do this. > > > > Is there anyway to remove these entries without restarting Asterisk? > > > > Any ideas on what could be done to prevent this? > > > > Example output: > > xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) > > No (d) Rx: BYE > > xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) > > No Rx: BYE > > xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) > > No Rx: BYE > > > > Thanks, > > Keith > > > > ___ > > -- 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
[asterisk-users] Disable SIP notify for peers
Hello, I am using OpenSER together with Asterisk. I have the users registered to OpenSER and have added peer definitions for each user so that the NOTIFY for MWI is sent to user when voicemail is left in their respective mailbox. That works great so far in terms of voicemail integration. On the OpenSER I have a script being executed for when message-summary SUBSCRIBE's are received which uses the manager interface for Asterisk to retrieve message counts and send them using sipsak. The one thing which I would like to change is that when I do a 'reload' or restart Asterisk, a NOTIFY is sent to each peer. When I have around 200 of these, Asterisk tries to send 200 NOTIFY messages at once which seems to sometimes lock it up and it also probably overloads the network unnecessarily. Is there any way to disable these NOTIFY's? I only want it to send them to the user when a voicemail is left. Thanks, Adrian ___ -- 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] Want to know Frequency and lenght of Frame
I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Thanks in advance. Regards, Sanjay. ___ -- 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] display time on Cisco 79xx
Chris Carey wrote: > They get the time from their NTP server > > On Mon, Mar 10, 2008 at 11:59 AM, Don Smith <[EMAIL PROTECTED]> wrote: > >> I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday >> Morning Daylight Savings time occurred. The server shows Mon Mar 10 >> 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show >> 09:59 10/03/08. How do I update the time display on the telephones please? Edit your SIPDefault.cnf file on your tftp server and do something like this: time_zone: EST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: March ; Month in which DST starts dst_start_day: "" ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 2 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Nov ; Month in which DST stops dst_stop_day: ""; Day of month in which DST stops dst_stop_day_of_week: Sunday; Day of week in which DST stops dst_stop_week_of_month: 1 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment -- Mark Johnson http://www.astroshapes.com/information-technology/blog/ ___ -- 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] Read function
Stupidity this is working
1,1,Answer()
1,n,Background(tt-weasels);
1,n,Read(CNT,,,2)
1,n,NoOP(${CNT})
if I wait when Background is timedout and then input digitst read
function receive inputed digits.
I think asterisk playing with me, AI rules :))
a little more and I will be in crazy house ^(
2008/3/10, Daniel Suleyman <[EMAIL PROTECTED]>:
> No other Soft phone doesn't helped, I tryed several codecs - same story :(.
>
>
> Where can be the problem?
>
> 2008/3/10, Daniel Suleyman <[EMAIL PROTECTED]>:
> > asterisk version 1.4.18
> > No I cant try hardfone but I can use other sip client, i'll chek it now
> >
> > 2008/3/10, Doug Lytle <[EMAIL PROTECTED]>:
> > > Daniel Suleyman wrote:
> > > > 2008/3/9, Doug Lytle <[EMAIL PROTECTED]>:
> > > >
> > > >> Daniel Suleyman wrote:
> > > >>
> > > >>> same story ^( no DTMF input
> > > >>>
> > > >>>
> > >
> > >
> > > What version of Asterisk?
> > > Can you try a different client, maybe even a SIP hard phone?
> > >
> > >
> > > 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
> > >
> >
>
___
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Re: [asterisk-users] Queue Pickup
>We have a customer service queue which works great. The members are hard >coded (member => SIP/1000), etc. However, we have a special need. If the >queue becomes busy, we would like to be able to dial an extension and >grab only the next caller in the queue. We don't want to log in as an >agent, since that would add another step (logging in/logging out). I saw >there was a Pickup() command, but I'm not sure if this will work with >queues. I have a "reverse transfer" module I wrote. I could probably adapt this for queues without too much work. Thoughts? Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Intermittent DTMF Problems
Lower the rxgain and txgain on your Zap channels. bilal ghayyad wrote: > Hi Brent; > > I have been suffering from this problem since about 2 > monthes and until now still did not resolved 100%. > > First of all, I need to tell u that mostly u have a > problem that the first digit is duplicated, for > example: if ur customer entered 108 then it will be > recognized 110 (the 1 duplicated, and then it takes > the 0, and it does not continue to take the 8 as it > completes the 3 digits ... this is just an example). > > Your problem is in the duplication for the digit and > specifically the first digit usually will be > duplicated. > > If u found a solution let me know. > > Regards > Bilal > > > --- > > I've recently installed Asterisk-based servers at > several of our branch > > offices. Each server has 2 X100P cards to handle 2 > incoming voice > lines. I was having a lot of trouble with Echo until > I got OSLEC > running on all of the servers, but now we have a new > problem. Incoming > > callers are not always able to dial extensions. I > would say probably > 95% of the calls go through correctly, but that other > 5% always get > dumped to the operator queue. I have relaxdtmf=yes in > my zapata.conf > for both channels, but it doesn't always help. One > particular customer > > has trouble dialing an extension about 1 call in 5. > I'm wondering if > it's just because they call us more than any of our > other customers or > if there is some peculiarity with their phone system. > Anybody have any > > ideas what to try next? > > Thanks, > Brent Davidson > > > > > > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > ___ > -- 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 > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Audiocodes MP124-FXS replying BUSY when line is not.
Hello,
Has anybody seen that Audiocodes gateway is replying with "486 Busy
here" when it's actually not (last call ended ~15 seconds ago).
I see this quite often. From other logs i see that previous call ends
at 11:13:01, then app_queue tries to dial at 11:13:14 and fails
numerous times, before succeeding at 11:14:02
I have attached sample SIP debug log:
Any ideas what i could try to change in config to avoid this? It's
config seems huge, maybe anybody has some experience with those
gateways?
Regards,
Atis
start of log
[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- Executing
[EMAIL PROTECTED]:70] Dial("Local/[EMAIL PROTECTED],2",
"SIP/90166|15|gtM(queue_call_answer^28254)") in new stack
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Insert SQL: INSERT INTO channels SET uniqueid = '1205172794.6453',
started = '1205172794', channel = 'SIP/90166-45079a0', state = 'Down',
callerid_num = '', callerid_name = '', accountcode = '', context =
'default-sip', exten = 's', priority = '1', application = '', data =
''
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: row
inserted on table: channels, id: 0
[Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166' is 1 out of 8
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Audio is at aa.bb.cc.dd port 47732
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x4 (ulaw) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x2 (gsm) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x8 (alaw) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x10 (g726aal2) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x20 (adpcm) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x40 (slin) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x80 (lpc10) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x400 (ilbc) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x800 (g726) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding non-codec 0x1
(telephone-event) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Reliably Transmitting (NAT)
to ee.ff.gg.hh:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-2067217913" ;tag=as18481a04
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Mar 2008 18:13:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471
v=0
o=root 31887 31887 IN IP4 aa.bb.cc.dd
s=session
c=IN IP4 aa.bb.cc.dd
t=0 0
m=audio 47732 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- Called 90166
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Update SQL: UPDATE channels SET callerid_num = '28254', callerid_name
= '', accountcode = '1205172743.6428', con
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Updated 1 rows on table: channels
<--- SIP read from ee.ff.gg.hh:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-2067217913" ;tag=as18481a04
To: ;tag=1c1673732975
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Supported: em,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004
Content-Length: 0
<->
[Mar 10 11:13:14] VERBOSE[31897] logger.c: --- (10 headers 0 lines) ---
[Mar 10 11:13:14] VERBOSE[31897] logger.c:
<--- SIP read from ee.ff.gg.hh:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-2067217913" ;tag=as18481a04
To: ;tag=1c1673732975
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact:
Supported: em,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004
Reason: Q.850 ;cause=17
Content-Length: 0
<->
[Mar 10 11:13:14] VERBOSE[31897] logger.c: --- (12 headers 0 lines) ---
[Mar 10 11:13:14] VERBOSE[31897] logger.c: -- Got SIP response 486
"Busy Here" back from ee.ff.gg.hh
[Mar 10 11:13:14] VERBOSE[31897] logger.c: Transmitting (NAT) to
ee.ff.gg.hh:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG
Re: [asterisk-users] display time on Cisco 79xx
Don Smith wrote: > > 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 > 10/03/08. How do I update the time display on the telephones please? > You'll need to edit the SIPDefault.cnf file. It'll be located in your TFTP directory. This is where you define the begin/end of DST. 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] display time on Cisco 79xx
They get UTC/GMT from the NTP server. It is up to the firmware on the phone to convert that date/time into the local time. No, it is not up to Asterisk, it is up to the phone firmware. Chris Carey wrote: > They get the time from their NTP server > > On Mon, Mar 10, 2008 at 11:59 AM, Don Smith <[EMAIL PROTECTED]> wrote: > >> I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday >> Morning Daylight Savings time occurred. The server shows Mon Mar 10 >> 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show >> 09:59 10/03/08. How do I update the time display on the telephones please? > > ___ > -- 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 > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Intermittent DTMF Problems
Hi Brent; I have been suffering from this problem since about 2 monthes and until now still did not resolved 100%. First of all, I need to tell u that mostly u have a problem that the first digit is duplicated, for example: if ur customer entered 108 then it will be recognized 110 (the 1 duplicated, and then it takes the 0, and it does not continue to take the 8 as it completes the 3 digits ... this is just an example). Your problem is in the duplication for the digit and specifically the first digit usually will be duplicated. If u found a solution let me know. Regards Bilal --- I've recently installed Asterisk-based servers at several of our branch offices. Each server has 2 X100P cards to handle 2 incoming voice lines. I was having a lot of trouble with Echo until I got OSLEC running on all of the servers, but now we have a new problem. Incoming callers are not always able to dial extensions. I would say probably 95% of the calls go through correctly, but that other 5% always get dumped to the operator queue. I have relaxdtmf=yes in my zapata.conf for both channels, but it doesn't always help. One particular customer has trouble dialing an extension about 1 call in 5. I'm wondering if it's just because they call us more than any of our other customers or if there is some peculiarity with their phone system. Anybody have any ideas what to try next? Thanks, Brent Davidson Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] display time on Cisco 79xx
They get the time from their NTP server On Mon, Mar 10, 2008 at 11:59 AM, Don Smith <[EMAIL PROTECTED]> wrote: > > I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday > Morning Daylight Savings time occurred. The server shows Mon Mar 10 > 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show > 09:59 10/03/08. How do I update the time display on the telephones please? ___ -- 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 1.6.0-beta5 Now Available
Could someone please update the links on asterisk.org to point to 1.6.0-beta5? They still point to 1.6.0-beta4, and beta 5 has been out for a few days now. Michael Cargile Software Developer Explido Software USA Inc. www.explido.us On Wed, 2008-03-05 at 15:50 -0600, The Asterisk Development Team wrote: > Greetings, > > The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of > this > beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug > fixes, the following new features have been added since beta4: > > * The SMDI interface in Asterisk has been reworked to fix a number of >issues as well as add some new features. SMDI message information >is now accessed in the dialplan using some new dialplan functions. >New options have been added to map Asterisk voicemail boxes to SMDI >station IDs. Also, MWI will now properly be sent for systems that have >some external interface modifying voicemail boxes, such as a web >interface, or with an email client in the case of IMAP storage. > > * The Postgres CDR module now supports some of the features of >cdr_adaptive_odbc. Specifically, you may add additional columns into >the table and they will be set, if you set the corresponding CDR >variable name. Also, if you omit columns in your database table, >those fields will be silently skipped when inserting the record. > > * The ResetCDR application now has an 'e' option that re-enables the >CDR if it has been disabled using the NoCDR option. > > * A new CLI command, "devstate change", has been added which allows you >to change the state of a Custom device. Custom device states were >previously only settable by using the DEVICE_STATE() dialplan function. > > * The Originate manager action now has its own permission level called >originate. Also, if you want this action to be able to execute > applications >that call out to a subshell, it requires the system privilege, as well. >These changes were made to enhance the security of the manager interface. > > For a full list of features that have been introduced from Asterisk 1.4 to > Asterisk 1.6.0, see the following file: > > * http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?view=markup > > For a full list of changes to Asterisk 1.6.0 from beta4 to beta5, see the > ChangeLog: > > * http://svn.digium.com/view/asterisk/tags/1.6.0-beta5/ChangeLog?view=markup > > There are a few more issues to resolve in 1.6.0 before it can enter release > candidate status, but we expect that to happen relatively soon. > > Thank you for your continued support of Asterisk! > > ___ > -- 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] display time on Cisco 79xx
On 10:59, Mon 10 Mar 08, Don Smith wrote: > I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday > Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 > PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 > 10/03/08. How do I update the time display on the telephones please? I guess they are not running Skinny right ? I have no idea how they work with the SIP image, but Skinny image gets the time from asterisk just fine -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called 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] Queue calls drop to voicemail intermittantly
so I tried this ...and locally on the same server, the channel variable 'VMFLAG' works great -- gets checked for direct calls, and set to 0 when sending calls through Queues. But the power of Chan Local/ to send calls between multiple servers is ruined because now if you dial direct a Local/ext ... it won't pass that variable along so all calls to external channels are considered Queue calls. :-\ gotta maintain that 'information' when I dial out over IAX2 to the other peer any random ideas appreciated, -- Chris "Chris Earle" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > To solve the problem of Local channels answering Queue calls, I thought > about myabe using a channel variable switch that turns on before the Queue > is called, and a check in the extension-dial to Local/ext to see if it is a > queue call and shouldn't go to voicemail, or if it's just a direct call that > should go to voicemail after a certain time. ...I think it'll work, > though I haven't tried it yet > > -- > > "Tilghman Lesher" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > > On Monday 17 December 2007 12:35, Gregory Malsack wrote: > > > Can anyone tell me what might cause callers on hold in a queue to drop > > > into agents voicemail boxes? > > > > Probably you're putting "Local" channels into the queue. Any answer event > at > > all generated by the Local channel, including one generated by Voicemail, > is > > considered a pickup by the Queue app. Note that if you use the raw > channel > > (SIP/IAX/Zap/whatever), then this will not happen when a queue member > fails to > > answer their phone. > > > > Or create extensions that do not end in Voicemail for the use of Queue. > > > > -- > > Tilghman > > > > ___ > > --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] Asterisk and Packetcable
Me too , I have one plataform with packet cable and I would like to implement con ASterisk > To: asterisk-users@lists.digium.com> From: [EMAIL PROTECTED]> Date: Mon, 10 > Mar 2008 17:10:29 +> Subject: Re: [asterisk-users] Asterisk and > Packetcable> > I am also interested in this.> > Sent from my Verizon Wireless > BlackBerry> > -Original Message-> From: "Carlos Alberto Bernat > Orozco" <[EMAIL PROTECTED]>> > Date: Mon, 10 Mar 2008 11:55:25 > > To:asterisk-users@lists.digium.com> Subject: [asterisk-users] Asterisk and > Packetcable> > > Hi group> > > I wrote 2 years ago to know if there is some > workaround for PacketCable. Since then I got no answer and now I hope there's > something about.> > Is there any chance to use Asterisk as softphone with > cable modem technology using Packetcable?> > Thanks in advanced> > > Carlos > Bernat> ___> -- 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 _ Tecnología, moda, motor, viajes,…suscríbete a nuestros boletines para estar siempre a la última http://newsletters.msn.com/hm/maintenanceeses.asp?L=ES&C=ES&P=WCMaintenance&Brand=WL&RU=http%3a%2f%2fmail.live.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] display time on Cisco 79xx
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 10/03/08. How do I update the time display on the telephones please? No virus found in this outgoing message. Checked by AVG. Version: 7.5.518 / Virus Database: 269.21.7/1323 - Release Date: 3/10/2008 11:07 AM ___ -- 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] Global Variables on Reload
Rob Schall wrote: > I'm running Asterisk 1.4.18 and having a problem with the > clearglobalvars option. > > I have a NIGHT_SERVICE variable which I initially set equal to off. I > then have an extension they can dial which will toggle that variable. My > problem is when you enter the CLI and type "reload", it resets to "off" > again. I've tried setting the clearglobalvars=no as well as just > commenting out that line, but no luck so far. > > Any ideas? we use MySQL db to store those global vars in our installation. i guess you can use and db to do that. -- Edwin Lam <[EMAIL PROTECTED]> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 ___ -- 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] Strange problem
Hi All, i'm experiencing a strange problem on sip channel. Sometime appens that the sip client ring as if it recieves 3 calls at the same time from the same number, even if thre is only a single call. I'm experiencing that both on the softphone sjphone and on the sip phone Grandstream GXP2000 an on two asterisk box one asterisk v 1.0.7 the second asterisk 1.2.16 I have not idea where to start for debug this Someone can help me? thank's in advance Accursio___ -- 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] NIN Ghosts music (free download) safe for MOH?
ASCAPs information is here: http://www.ascap.com/licensing/generalreports.html BMIs information is here: http://www.bmi.com/licensing/?link=navbar ...brig Brig C. McCoy Network Administrator ThyssenKrupp Access Corp 4001 E 138th ST Grandview, MO 64030 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Kinard Sent: Monday, March 10, 2008 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam:Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH? -Original Message- From: [EMAIL PROTECTED] On Behalf Of John Faubion > Ok now I am curious, if a radio is playing in a store, a restaurant or at > the beach, wouldn't that be considered a public performance? And even though > the radio station has already paid the license fee, does this mean that the > person who owns the radio is also subject to these fees? I know of several > key systems with FM radio cards providing MoH and I've often wondered about > the ramifications of that setup and the music industry. Well, ASCAP/BMI are stingy on collecting their fees -- one of them even went after a shop for playing the Monday Night Football theme just because they had the TV on that channel when it came on. It seems kind of ruthless if you ask me, but these guys play their cards well, and generally avoid attracting the same the kind of infamy that the RIAA has managed to garner. Unlike the RIAA, they actually pay out their collected royalties. I don't know what their fees even are, but I think they're not too bad from a business' standpoint. Probably can't hurt to call them up and just ask. They might be willing to explain things in better detail. Just be wary if they want your company's information :) --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 As you are aware, messages sent by e-mail can be manipulated by third parties. For this reason our e-mail messages are usually not legally binding. This electronic message (including any attachments) contains confidential information and may be privileged or otherwise protected from disclosure. The information is intended to be for the use of the intended addressee only. Please be aware that any disclosure, copy, distribution or use of the contents of this message is prohibited. If you have received this e-mail in error please notify me immediately by reply e-mail and delete this message and any attachments from your system. Thank you for your cooperation. ___ -- 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] NIN Ghosts music (free download) safe for MOH?
-Original Message- From: [EMAIL PROTECTED] On Behalf Of John Faubion > Ok now I am curious, if a radio is playing in a store, a restaurant or at > the beach, wouldn't that be considered a public performance? And even though > the radio station has already paid the license fee, does this mean that the > person who owns the radio is also subject to these fees? I know of several > key systems with FM radio cards providing MoH and I've often wondered about > the ramifications of that setup and the music industry. Well, ASCAP/BMI are stingy on collecting their fees -- one of them even went after a shop for playing the Monday Night Football theme just because they had the TV on that channel when it came on. It seems kind of ruthless if you ask me, but these guys play their cards well, and generally avoid attracting the same the kind of infamy that the RIAA has managed to garner. Unlike the RIAA, they actually pay out their collected royalties. I don't know what their fees even are, but I think they're not too bad from a business' standpoint. Probably can't hurt to call them up and just ask. They might be willing to explain things in better detail. Just be wary if they want your company's information :) --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
Re: [asterisk-users] Asterisk and Packetcable
I am also interested in this. Sent from my Verizon Wireless BlackBerry -Original Message- From: "Carlos Alberto Bernat Orozco" <[EMAIL PROTECTED]> Date: Mon, 10 Mar 2008 11:55:25 To:asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Packetcable Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat ___ -- 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] Redirecting channels?
Hello I am going to have a setup like this: One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the other hand, I also have another box with VoiceGuide and Dialogic. As a temporary migration-solution i would like to redirect some of the ISDN30 channels from the Asterisk to the Dialogic-box. How would I do this? ___ -- 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 and Packetcable
Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat ___ -- 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] NIN Ghosts music (free download) safe for MOH?
Hi, I sense a confusion here between two things, On Mon, Mar 10, 2008 at 11:25:22AM -0400, John Novack wrote: > > > Horwich IT Services (Godwin Stewart) wrote: > > I lived there from 1983 until a few months ago and I know for a fact > > that bars have to have special TV licenses in order to show, for > > example, soccer matches and other sporting events, and a radio > > license in order to broadcast the radio to clients, many of whom > > are too p*ssed to realize what they're listening to or watching anyway :) We all seem to feel that this "is just not right". But what is the issue here specifically? The issue is the limits of fair use. I can read a book whereever I want. I can put a record on loud speakers at my house The neighbourghrs might complain, but not the record companies. Public performance has been given a different status and is goverened by somehwat different rules. E.g. for the case of the barber playing a radio in his shop you can say that he uses the music for commercial purpose, and OTOH, the owners of the radio station have already paid for this, so why pay twice, etc. etc. . Well, this might be an interesting topic for rants. But I believe that the legal opnions are generally quite clear on that playing on-hold music to entartiain the folks waiting on your line is a sort of public performance. You cannot get off the hook with any "fair use" clause. > > > Certainly the case in the US as well. ASCAP goes on regular campaigns > with Pizza shops and the like. Look for the yellow sign on the door. > So far they haven't bothered medical offices too much, and I do not know > about XM radio sold for commercial use. I suspect that MAY be covered. > > Bottom line is if you write it your self, and play it your self, in the > US you probably will be OK. Other than that, you have exposure, or you > AND your client . You can look at them as being extortionists. You can just pay them and make them go away. Well, if that were the case, other alternatives were not possible. FreePlay Music was mentioned up this thread. Just as much as some prompts sets available for Asterisk are perfectly legal for usage in this sort of "public performance" (IVR system), sound files released under a permissive enough license allow it. But you have goons^Wlawyers to answer to. So you have to keep yourself covered: make sure you don't just pick a sound file from somewhere in the internet. Make sure you can track it to its copyrights holder and that it is accompanied by a clear license. And on the flip side: if you release your works, please put a clear license next to it, or at least a reference to one. It is frustrating to see some useful stuff on the internet that you their author would love other to use. But you just can't be sure of that, because there's not a word about the license. -- 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
[asterisk-users] Queue Pickup
Running Asterisk 1.4... We have a customer service queue which works great. The members are hard coded (member => SIP/1000), etc. However, we have a special need. If the queue becomes busy, we would like to be able to dial an extension and grab only the next caller in the queue. We don't want to log in as an agent, since that would add another step (logging in/logging out). I saw there was a Pickup() command, but I'm not sure if this will work with queues. Any thoughts? Rob ___ -- 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] Read function
No other Soft phone doesn't helped, I tryed several codecs - same story :(. Where can be the problem? 2008/3/10, Daniel Suleyman <[EMAIL PROTECTED]>: > asterisk version 1.4.18 > No I cant try hardfone but I can use other sip client, i'll chek it now > > 2008/3/10, Doug Lytle <[EMAIL PROTECTED]>: > > Daniel Suleyman wrote: > > > 2008/3/9, Doug Lytle <[EMAIL PROTECTED]>: > > > > > >> Daniel Suleyman wrote: > > >> > > >>> same story ^( no DTMF input > > >>> > > >>> > > > > > > What version of Asterisk? > > Can you try a different client, maybe even a SIP hard phone? > > > > > > 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 > > > ___ -- 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] Intermittent DTMF Problems
I've recently installed Asterisk-based servers at several of our branch offices. Each server has 2 X100P cards to handle 2 incoming voice lines. I was having a lot of trouble with Echo until I got OSLEC running on all of the servers, but now we have a new problem. Incoming callers are not always able to dial extensions. I would say probably 95% of the calls go through correctly, but that other 5% always get dumped to the operator queue. I have relaxdtmf=yes in my zapata.conf for both channels, but it doesn't always help. One particular customer has trouble dialing an extension about 1 call in 5. I'm wondering if it's just because they call us more than any of our other customers or if there is some peculiarity with their phone system. Anybody have any ideas what to try next? Thanks, Brent Davidson ___ -- 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] NIN Ghosts music (free download) safe for MOH?
Horwich IT Services (Godwin Stewart) wrote: > On Mon, 10 Mar 2008 09:17:16 +0100, Dave Cotton > <[EMAIL PROTECTED]> wrote: > > >>> Ok now I am curious, if a radio is playing in a store, a restaurant or at >>> the beach, wouldn't that be considered a public performance? >>> >> From a conversation with a hairdresser who fell foul of this the answer is >> in France you do have to pay. >> > > Confirmed. > > I lived there from 1983 until a few months ago and I know for a fact that > bars have to have special TV licenses in order to show, for example, soccer > matches and other sporting events, and a radio license in order to broadcast > the radio to clients, many of whom are too p*ssed to realize what they're > listening to or watching anyway :) > Certainly the case in the US as well. ASCAP goes on regular campaigns with Pizza shops and the like. Look for the yellow sign on the door. So far they haven't bothered medical offices too much, and I do not know about XM radio sold for commercial use. I suspect that MAY be covered. Bottom line is if you write it your self, and play it your self, in the US you probably will be OK. Other than that, you have exposure, or you AND your client . John Novack -- Dog is my co-pilot ___ -- 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] Global Variables on Reload
On Mon, Mar 10, 2008 at 09:09:41AM -0500, Rob Schall wrote: > I'm running Asterisk 1.4.18 and having a problem with the > clearglobalvars option. > > I have a NIGHT_SERVICE variable which I initially set equal to off. I > then have an extension they can dial which will toggle that variable. My > problem is when you enter the CLI and type "reload", it resets to "off" > again. I've tried setting the clearglobalvars=no as well as just > commenting out that line, but no luck so far. > > Any ideas? What happens if Asterisk is restarted? Use the DB? -- 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
[asterisk-users] Shared Extension
I am working on a project that requires shared extension. Where shared line looks at the status of a line/trunk, shared extension would look at a series of channels as the same "extension". The users would like to add destination channels on the fly, to provide roaming extensions, but maintaining fixed channels as well. If a call comes in on an extension, the system needs to honor the fact that channel 1 is busy, therefore, the extension is busy. Keep in mind that the channel could be anything including SIP outbound trunk channels (read cell phone or hotel room). The Dial command does provide a nice multi-channel dialer, especially with the "r" option, however, if one of the lines is busy, the system will keep ringing the other lines until timeout or answer (read voice mail). So I am contemplating adding a feature to the dial command, that would make any channel busy, cause the initial Dial to come back as busy. Kind of a force the state flag. Before I brake into code, does anyone have any other ideas? This would also help with phones like Grandstream, where you have 4 accounts to configure, and would like to have all 4 SIP accounts act as 1 extension. Tony Plack ___ -- 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] Call forwarding-in india
On Mon, 10 Mar 2008 16:22:45 +0530, "sandeep" <[EMAIL PROTECTED]> wrote: > Can any body tell how to enable call forward facility in INDAI > for an asterisk IPPBX. Why would it be different in India from anywhere else? -- Godwin Stewart - Horwich IT services ___ -- 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] Had it with Dell Garbage
I had some problems with tyan mobos (digium hardware incompatible) 2008/3/10, Matt Riddell <[EMAIL PROTECTED]>: > > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Mike Trest - Personal wrote: > > Steve, > > I have fielded several hundred Asterisk and related VoIP boxes. > > I buy SuperMicro 1-U units mostly. I have also used their larger > > units with RAID and a full load of ULTRA SCSI (for MySql application). > > I'd second the recommendation on SuperMicro - had nothing but goodness > from them. > > - -- > Kind Regards, > > Matt Riddell > Director > ___ > > http://www.venturevoip.com (Great new VoIP end to end solution) > http://www.venturevoip.com/news.php (Daily Asterisk News - html) > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.7 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFH1KjHDQNt8rg0Kp4RAvLyAJkBpmdfD1zuDzGnDMlODVmVI7vfTgCeI0WI > BCpHmKZc15z+ZBcoYLm75a4= > =PVmD > -END PGP 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 > ___ -- 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] Microsoft Office Communications Server
They very likely purchased or licensed an engine from someone. Use Wireshark and compare it to other SIP proxies/servers/gateways. On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Has anyone done any integration with this? > > All I know so far is that it appears to use some non standard form of SIP. > > Any pointers? > > - -- > Kind Regards, > > Matt Riddell > Director > ___ > > http://www.venturevoip.com (Great new VoIP end to end solution) > http://www.venturevoip.com/news.php (Daily Asterisk News - html) > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.7 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFH1KVoDQNt8rg0Kp4RAgSfAJ0aXGIkKi6kGAjZK8TtSV2mMj79qQCdHhAS > 1jZ9sjtsTJ3O1R9J3giztw8= > =Mlnt > -END PGP 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 > -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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] Global Variables on Reload
I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just commenting out that line, but no luck so far. Any ideas? Rob ___ -- 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] Call forwarding-in india
well give us details 2008/3/10, sandeep <[EMAIL PROTECTED]>: > > Hi All, > Can any body tell how to enable call forward facility in INDAI > for an asterisk IPPBX. > > Regards, > Sandeep.S > > ___ > -- 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] FaxBack Service with Asterisk
Hi, Has anyone ever used asterisk for a faxback service ? Thanks. Dovid___ -- 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] Dead Air on PF firewall
On 07:00, Mon 10 Mar 08, NOC ph wrote:
> Hi All,
>
> I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
> can make a call but some reasons I have a dead air.
>
> Any Ideas? below are my rules...
>
> ext_if = "bce0"
> int_if = "bce1"
> altitude = "172.16.1.0/24"
>
> machines
> vbox = "172.16.1.1"
> uci = "172.16.1.4"
> voices = "203.172.x.1"
> ipc = "203.172.x.2"
>
> default deny
> set block-policy return
> set loginterface $ext_if
> set skip on lo
> scrub in
>
> nat
> nat on $ext_if from !($ext_if) -> ($ext_if:0)
> nat on $ext_if inet proto { udp tcp } from $vbox to any port 5060 ->
> $ext_if port 5060
> nat on $ext_if inet proto tcp from $uci to any port 1500 -> $ext_if port
> 1500
Why those two rules ? The first nat rule already takes care
of that
> rdr on $ext_if proto { udp tcp } from any to $ext_if port 5060 -> $vbox
> port 5060
> rdr on $ext_if proto udp from any to $ext_if port 5100 -> $vbox port 5100
you have to forward the rtp ports as well
rdr on $ext_if proto udp from any to $ext_if port
1:2 -> $vbox
>
> filtering section
> pass out on { $int_if, ext_if } inet proto { udp tcp } from $altitude to any
> pass in on $ext_if inet proto { tcp udp } from $ipc to any port 5060
> pass in on $ext_if inet proto tcp from $ipc to any port 1500 flags S/SA
> keep state
And you should allow the rtp ports as well
pass in on $ext_if inet proto udp from any to any port
1:2 keep state
> pass in on bce0 proto tcp from $ipc to any port ssh flags S/SA keep state
> pass in inet proto icmp all icmp-type echoreq keep state
> pass in quick on bce1
>
For reference, here are my pf rules for my internal pbx:
##
# Macros #
##
ext_if = "rl0"
ext_ip = "82.95.XXX.XXX"
int_if = "wb0"
int_net = "192.168.2.0/24"
voip_server = "192.168.2.4"
voip_ports = "{ 4569, 5060, 1:2 }"
# NAT rules: "rdr", "nat", "binat" #
nat on $ext_if from $int_if:network to any -> $ext_ip
# asterisk server
rdr on $ext_if proto udp from any to any port $voip_ports ->
$voip_server
#
# Filtering #
#
# voip always goes in the priority class
pass out quick on $ext_if inet proto udp from any to any
port $voip_ports keep state queue q_pri
pass in quick on $ext_if inet proto udp from any to any port
$voip_ports keep state queue q_pri
Also, make sure in asterisk sip.conf you have the externip
and localnet config parameters set.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
"Why is it drug addicts and computer aficionados are both called users?"
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Re: [asterisk-users] Microsoft Office Communications Server
> > Has anyone done any integration with this? > > All I know so far is that it appears to use some non standard form of > SIP. > > Any pointers? > What!? Microsoft implementing something not compliant with official standards. Your kidding? Sorry Matt, no advice here but I just couldn't resist. -- David Cook ___ -- 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] Dead Air on PF firewall
On Mon, 10 Mar 2008 07:00:17 +0800, NOC ph <[EMAIL PROTECTED]> wrote: > I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I > can make a call but some reasons I have a dead air. Judging by the fact that you're portforwarding port 5060, I'm guessing that you're using SIP with the outside. This also means that you need to allow the RTP stream though your NAT FW. Port 5060 only carries the signalling, the audio is carried by the RTP stream, which is why you're getting no audio. Google will probably let you know which UDP ports your appliances are using for the RTP stream. General help that you'll be able to refine WRT the specifics of your setup is available here: http://www.google.com/search?q=asterisk+%22no+audio%22 -- Godwin Stewart - Horwich IT services ___ -- 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] dialstatus and cancelled calls
According to http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS when a caller hangs up before the callee has time to pick the phone up then DIALSTATUS should be CANCEL. And it is. However, the disposition field in the CDR table is "NO ANSWER". So if I analyze the CDR data I won't be able to discriminate calls cancelled by the caller and calls not answered by the callee (timeout). I get the same disposition value whether I use cdr-csv or MySQL via asterisk-addons. I'm using * 1.2.26.2. How can I get the DIALSTATUS value to the disposition field? Would I have to do it manually in my dialplan via Set(CDR(disposition))? Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] Dead Air on PF firewall
Hi All,
I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I
can make a call but some reasons I have a dead air.
Any Ideas? below are my rules...
ext_if = "bce0"
int_if = "bce1"
altitude = "172.16.1.0/24"
machines
vbox = "172.16.1.1"
uci = "172.16.1.4"
voices = "203.172.x.1"
ipc = "203.172.x.2"
default deny
set block-policy return
set loginterface $ext_if
set skip on lo
scrub in
nat
nat on $ext_if from !($ext_if) -> ($ext_if:0)
nat on $ext_if inet proto { udp tcp } from $vbox to any port 5060 ->
$ext_if port 5060
nat on $ext_if inet proto tcp from $uci to any port 1500 -> $ext_if port
1500
rdr on $ext_if proto { udp tcp } from any to $ext_if port 5060 -> $vbox
port 5060
rdr on $ext_if proto udp from any to $ext_if port 5100 -> $vbox port 5100
filtering section
pass out on { $int_if, ext_if } inet proto { udp tcp } from $altitude to any
pass in on $ext_if inet proto { tcp udp } from $ipc to any port 5060
pass in on $ext_if inet proto tcp from $ipc to any port 1500 flags S/SA
keep state
pass in on bce0 proto tcp from $ipc to any port ssh flags S/SA keep state
pass in inet proto icmp all icmp-type echoreq keep state
pass in quick on bce1
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[asterisk-users] call forward facility in INDIA
Hi All, Can any body tell how to enable call forward facility in INDIA for an asterisk IPPBX. Regards, Sandeep.S___ -- 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] Call forwarding-in india
Hi All, Can any body tell how to enable call forward facility in INDAI for an asterisk IPPBX. Regards, Sandeep.S___ -- 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] dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Iedema wrote: > On 3/10/08, Michael Iedema <[EMAIL PROTECTED]> wrote: >> Hello everyone, >> >> I'm having some troubles with some dialplan logic I've written which >> sends missed call notifications via e-mail. It's currently sending >> these notifications even if the call was answered, marking them all as >> hung-up. What I've been able to see is that the macro never reaches >> the "s-ANSWER" bits which mark the call as successful. >> >> I've posted my extensions.conf and a call trace to pastebin[1]. The >> extensions.conf may look a bit funny as it is the internally generated >> file from a project I'm working on. There are enough comments in there >> to make it human readable. >> >> All feedback is appreciated. >> >> Regards, >> -Michael >> >> [1] http://pastebin.ca/936296 >> > > Sorry for the noise, I've figured it out. The priorities after the > Dial() are not executed if the call was completed. I now handle the > ANSWER status detection in the h extension. Or use the "g" option to the dial command: g- Proceed with dialplan execution at the current extension if the destination channel hangs up. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1Q/3DQNt8rg0Kp4RAq84AKC2uIDQmVI/Y8OyrzXbO4jqsL794wCeIPEn P3ZdCTJeA7BPfde4EaoNrZc= =5jey -END PGP 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
Re: [asterisk-users] 1.6.beta5 (format 0x40 (slin))
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: > exten => s,2,BackGround(/var/lib/asterisk/sounds/en/vm-instructions.gsm) Drop the .gsm at the end of the filename. Asterisk will chose the best format for the call. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1Q9qDQNt8rg0Kp4RAvGJAJ9n6od52e3URmdzIAM2ApAJ3hoWngCfRktZ IlLrxWu5pPzkSa6R84AmRkQ= =i4jr -END PGP 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
Re: [asterisk-users] dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)
On 3/10/08, Michael Iedema <[EMAIL PROTECTED]> wrote: > Hello everyone, > > I'm having some troubles with some dialplan logic I've written which > sends missed call notifications via e-mail. It's currently sending > these notifications even if the call was answered, marking them all as > hung-up. What I've been able to see is that the macro never reaches > the "s-ANSWER" bits which mark the call as successful. > > I've posted my extensions.conf and a call trace to pastebin[1]. The > extensions.conf may look a bit funny as it is the internally generated > file from a project I'm working on. There are enough comments in there > to make it human readable. > > All feedback is appreciated. > > Regards, > -Michael > > [1] http://pastebin.ca/936296 > Sorry for the noise, I've figured it out. The priorities after the Dial() are not executed if the call was completed. I now handle the ANSWER status detection in the h extension. -Michael ___ -- 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 logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)
Hello everyone, I'm having some troubles with some dialplan logic I've written which sends missed call notifications via e-mail. It's currently sending these notifications even if the call was answered, marking them all as hung-up. What I've been able to see is that the macro never reaches the "s-ANSWER" bits which mark the call as successful. I've posted my extensions.conf and a call trace to pastebin[1]. The extensions.conf may look a bit funny as it is the internally generated file from a project I'm working on. There are enough comments in there to make it human readable. All feedback is appreciated. Regards, -Michael [1] http://pastebin.ca/936296 ___ -- 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] NIN Ghosts music (free download) safe for MOH?
AFAIK (as a member of SACEM and BMI), anyone who uses music in any commercial context like a store open to the public must pay royalties on it *if* the music is registered via ASCAP, BMI, SACEM or some other rights collection organization. This is usually done on a yearly basis. Those that use "storecast" subcarrier feeds (does that even still exist?) have the royalties included. You might find some music of use here: http://mediaminutes.net/music/music.rss One or two may be suitable for moh. I know you can use them for anything you like. /r ___ -- 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] NIN Ghosts music (free download) safe for MOH?
Am Montag, den 10.03.2008, 02:59 -0500 schrieb John Faubion: > > But, just to clarify, please remember that using music as MoH > > is considered a "public performance", and if the pieces in > > question do not include a buyout license *for the performance > > Ok now I am curious, if a radio is playing in a store, a restaurant or at > the beach, wouldn't that be considered a public performance? And even though > the radio station has already paid the license fee, does this mean that the > person who owns the radio is also subject to these fees? I know of several > key systems with FM radio cards providing MoH and I've often wondered about > the ramifications of that setup and the music industry. Good morning, the legal situation probably differs between countries. In Germany, you are required to register with the GEMA if you intend to play music in public if the artist is a GEMA customer. If you _only_ play free music, the law does not require you to register afaik, but in doubt you will have to prove that you did not play GEMA music (which is ridiculous when you think about it, but you do not want to fight against that machine). A party where two guests do not know each other's names may be considered public, even if only ten or twenty people are there. A class room, a barber shop, a supermarket or having a barbecue on the beach are surely public. The fees due will be calculated in regard to the area where the event takes place, because that limits the _maximum_ audience. Ain't it nice. (No idea though how exactly the area for music on hold is calculated - have a look at their tariffs jungle at http://www.gema.de/musiknutzer/abspielen-auffuehren/tarife-im-ueberblick/ ). I am not a lawyer, and am still lucky to not have to do with those music industry guys (and who is the "pirate" here...). BR Anselm ___ -- 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] NIN Ghosts music (free download) safe for MOH?
John Faubion wrote: >> But, just to clarify, please remember that using music as MoH >> is considered a "public performance", and if the pieces in >> question do not include a buyout license *for the performance > > Ok now I am curious, if a radio is playing in a store, a restaurant or at > the beach, wouldn't that be considered a public performance? And even though > the radio station has already paid the license fee, does this mean that the > person who owns the radio is also subject to these fees? From a conversation with a hairdresser who fell foul of this the answer is in France you do have to pay. Dave Cotton ___ -- 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] NIN Ghosts music (free download) safe for MOH?
On Mon, 10 Mar 2008 09:17:16 +0100, Dave Cotton <[EMAIL PROTECTED]> wrote: > > Ok now I am curious, if a radio is playing in a store, a restaurant or > > at the beach, wouldn't that be considered a public performance? > > From a conversation with a hairdresser who fell foul of this the answer > is in France you do have to pay. Confirmed. I lived there from 1983 until a few months ago and I know for a fact that bars have to have special TV licenses in order to show, for example, soccer matches and other sporting events, and a radio license in order to broadcast the radio to clients, many of whom are too p*ssed to realize what they're listening to or watching anyway :) -- Godwin Stewart - Horwich IT services ___ -- 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] NIN Ghosts music (free download) safe for MOH?
On the subject of hold music, I've been using stuff from stock20.com. They've got a good selection and they only charge $7 per song, and you can do anything you like with it. I did my own voiceovers (I built a very bad "isolation booth" in my basement using blankets and wood clamps. I wish I was making that up) and saved a bunch of money over what some companies charge for that sort of thing. ___ -- 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] Fwd: {s} - extension
thanks. 2008/3/10, Noah Miller <[EMAIL PROTECTED]>: > Hi Daniel - > > > Thank you for guide most things become cleare. No I dont need the dial tone. > > When I pickup XLITE to dial a number I hear dialtone and after I enter > > number nothing happens, this behaviar was strange for me, exactly > > becase you said I have analog phone in mind :) > > The only thing you need is to have XLite and a matching extension for > the number you want to dial in the same context, or in an included > context. If you do that (and your dial() statement is correct), it > will work. > > - Noah > > > > On Sun, Mar 9, 2008 at 12:35 PM, Daniel Suleyman <[EMAIL PROTECTED]> wrote: > > Thank you for guide most things become cleare. No I dont need the dial tone. > > When I pickup XLITE to dial a number I hear dialtone and after I enter > > number nothing happens, this behaviar was strange for me, exactly > > becase you said I have analog phone in mind :) > > > > > > 2008/3/9, Tzafrir Cohen <[EMAIL PROTECTED]>: > > > > > > > On Sun, Mar 09, 2008 at 10:19:05AM +0400, Daniel Suleyman wrote: > > > > ok, then I'm not understanding something. > > > > How I can call with xlite to my Asterisk not sending extension? > > > > when I want to call I pick up phone, hear ring (piii) and I need > > > > to type some extension otherwise nothing hapens > > > > > > You have an analog phone in mind. In most other cases the "dialtone" is > > > produced by a device other than the PBX. > > > > > > Why exactly do you need that dialtone? Why not just send a number? > > > > > > -- > > > 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 > > > > > > > ___ > > -- 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] Read function
asterisk version 1.4.18 No I cant try hardfone but I can use other sip client, i'll chek it now 2008/3/10, Doug Lytle <[EMAIL PROTECTED]>: > Daniel Suleyman wrote: > > 2008/3/9, Doug Lytle <[EMAIL PROTECTED]>: > > > >> Daniel Suleyman wrote: > >> > >>> same story ^( no DTMF input > >>> > >>> > > > What version of Asterisk? > Can you try a different client, maybe even a SIP hard phone? > > > 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 > ___ -- 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] replace astdb with a cluster-capable sql database engine
--- Vieri <[EMAIL PROTECTED]> wrote: > Would it be possible to modify the API calls that > are > currently going to the AstDB code within Asterisk, > and > put a translation layer to have them use the > func_odbc > instead (or either one)? > At a lower level, for everything Asterisk does to > its > AstDB, maybe there could be a system setting which > allows the user to say, ok, use DB, or, no, use > func_odbc (not at the dialplan level). What I mean is that ast_db_put and similar calls are about everywhere within the 1.2 base code (eg. chan_zap, chan_sip, chan_iax2, pbx_dundi, etc). There are a lot of applications out there (not just easily modifiable dialplans) that make use of the DB calls (custom add on code, XML phone applications, etc). So putting a translation layer so that ast_db_* API calls either go the normal route or translate to func_odbc (or another path) would improve functionality because both old and new apps would be able to seamlessly take advantage of the new database backend or keep using DB1 (the * admin would decide). I haven't looked at the 1.4/1.6 source code yet but I was wondering how many people would benefit from this. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] NIN Ghosts music (free download) safe for MOH?
> But, just to clarify, please remember that using music as MoH > is considered a "public performance", and if the pieces in > question do not include a buyout license *for the performance Ok now I am curious, if a radio is playing in a store, a restaurant or at the beach, wouldn't that be considered a public performance? And even though the radio station has already paid the license fee, does this mean that the person who owns the radio is also subject to these fees? I know of several key systems with FM radio cards providing MoH and I've often wondered about the ramifications of that setup and the music industry. John Faubion ___ -- 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
