SV: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail
As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that in Gentoo I would really appreciate it. Thanks -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Tomislav Parcina Sendt: 22. februar 2006 08:52 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: Fromstring when sending e-mail on recievedvoicemail In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always [EMAIL PROTECTED] and the name of the sender is always Added by portage for asterisk. I want to change both sender-address and the name of the sender. In voicemail.conf you have [general] [EMAIL PROTECTED] fromstring=My name -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hints
Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
Have you checked the permissions on the file? Is it executable? Garth Dirgan Putra wrote: hi All need help, iam installing areskiCC and have a problem after that create extension for calling card and after dial exten = 17000,3,DeadAgi,a2billing.php i see messages : a2billing.php no such file in directory, i tired copy that file that file aready copy in agi-bin. any body have experience in same problem, i need a suggestion to solve this problme thanks Putra Do you Yahoo!? Yahoo! Movies http://sg.rd.yahoo.com/mail/sg/footer/def/*http://sg.movies.yahoo.com - Search movie info and celeb profiles and photos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call queue design issues and suggestions
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know if this works for you, but I use the following mechanism. I don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff. For each queue, dialing the extension (), puts the caller into the queue (ie, a customer calling for reservations). I use ** to sign a phone into the queue and * to sign out of a queue. Good idea, maybe sometimes I'll need it. You can use the manager to see who is currently logged into a port. It doesn't take much to write a cgi script that outputs the Cisco XML for the phones. I've built a few apps that do interesting things. It would be quite easy to write an app that: It could be easy for someone with experience, but if you have never done it before (like me) it isn't like that. Can you send us what you have done? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot see the caller id , When calls made from one server to another
Hi
I had installed and configured 2 IAX server ,
users from 1'st server can dial to the second server
and vice versa
But when I make calls to users in other
server , on my client , I get the caller if as
[EMAIL PROTECTED] , the same I get when I try
reverse , ie I get on my cleint caller id as
[EMAIL PROTECTED]
Please guide me what should I do
for displaying the user id when users from one
server calls to other server users [ I get the
call-ID when users dial to the same server they are ]
part of my iax.conf file is as follows
(20.32)
[johnb]
type=friend
user=johna
secret=secret
host=192.168.20.99
context=project
extensions.conf (20.32)
exten = _4XXX,1,Dial(IAX2/johnb/${EXTEN:1},30,r)
exten = _4XXX,2,Congestion
my iax.conf file in (20.99) is
[johna]
type=friend
user=johna
secret=secret
host=192.168.20.32
context=project
extensions.conf (20.99) contains
exten = _3XXX,1,Dial(IAX2/johna/${EXTEN:1},30,r)
exten = _3XXX,2,Congestion
___
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[Asterisk-Users] Re: Send flash through zap channel
OR you can try this: in features.conf: [applicationmap] zapflash=*3,callee,flash if you put any spaces in the above line, it will not work!!! in extensions.conf add this line right before the dial commands where you want this to work: exten = s,12, set(DYNAMIC_FEATURES=zapflash) Then *3 should flash the line. Thanx, so far this was what I missed. But ... Feb 22 09:27:13 WARNING[28084]: app_flash.c:101 flash_exec: Zap/1-1 is not an FXO Channel -- Hungup 'Zap/1-1' So the ISDN Bri (qozap) seems not to support flash. But hey, with an ISDN-Phone directly connected to the NEC PBX it works, so there seems to be a flash-equivalent in ISDN signalisation. Anyone got a clue how to signal flash over zaptel ISDN trunk? Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Any news about new Snom 300? Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot see the caller id , When calls made from one server to another
John Joseph wrote:
Hi
I had installed and configured 2 IAX server ,
users from 1'st server can dial to the second server
and vice versa
But when I make calls to users in other
server , on my client , I get the caller if as
[EMAIL PROTECTED] , the same I get when I try
reverse , ie I get on my cleint caller id as
[EMAIL PROTECTED]
Please guide me what should I do
for displaying the user id when users from one
server calls to other server users [ I get the
call-ID when users dial to the same server they are ]
part of my iax.conf file is as follows
(20.32)
[johnb]
type=friend
user=johna
secret=secret
host=192.168.20.99
context=project
extensions.conf (20.32)
exten = _4XXX,1,Dial(IAX2/johnb/${EXTEN:1},30,r)
exten = _4XXX,2,Congestion
my iax.conf file in (20.99) is
[johna]
type=friend
user=johna
secret=secret
host=192.168.20.32
context=project
extensions.conf (20.99) contains
exten = _3XXX,1,Dial(IAX2/johna/${EXTEN:1},30,r)
exten = _3XXX,2,Congestion
John,
in iax.conf, there are settings for callerid,
so try callerid=johna
or callerid=asrecieved
and usecallerid=yes
should do it
yusuf
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Re: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
My telephone extensions on asterisk which itself is connected to the Bell line using SPA-3000, ring only after third ring from the caller. Why is this happening and what is the solution? First, the ringback tone that a caller receives is not in sync with the actual phone line ringing. In some cases, the ringback tone is almost (but not quit) one ring cycle prior to the actual phone line ringing. Second, in the US the spa3k itself needs wait for the callerid to complete _before_ it sends anything to asterisk. Callerid happens between the first and second ring as others have already noted. Depending on the ringback tone sync noted above and waiting for the callerid, that elapsed time can approach what would be considered the third ring. (You can watch asterisk's CLI for the incoming call and judge the delay incurred within the spa3k.) Once asterisk receives the sip exchange from the spa3k, asterisk runs through your dialplan, and sends an appropriate sip packet to your sip phones to cause it to ring. There could be some delay in the phone actually ringing, and that delay will be 99% the result of how the sip phone manufacturer handles the ring packet. Watching the CLI and listening for the ring will provide some level of indication as to how much delay is associated with it. You might also trying experimenting with some of the spa3k parameters. If I recall correctly, there is a configurable parameter associated with how many rings have to occur before the spa3k forwards the sip packets to asterisk. Its been a while since I've messed with the spa3k so not sure where its located in their menues. As others have already mentioned, if you don't use/have callerid on your pstn line, then disable callerid to expedite the call progress. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
need help, iam installing areskiCC and have a problem after that create extension for calling card and after dial exten = 17000,3,DeadAgi,a2billing.php i see messages : a2billing.php no such file in directory, i tired copy that file that file aready copy in agi-bin. Try chmod 755 /var/lib/asterisk/agi-bin/a2billing.php benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] did from sip trunk
I want to do inbound routing of calls comming from sip trunks. Is there a way to force the DID that comes from a trunk that does not have DID support? (something like using the outgoing caller-id for the trunk?) My problem is this: I've got several sip trunks (SPA3000). I want to have an IVR in all but one of them, the one that is connected to a cellular adapter. In this line I want to let it ring until somebody picks up because many times we checks the caller-id and calls him back. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 79xx firmware
I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me technical support agreement which cost around 75$ for every phone! -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma A200D analog card with fxo's
The limited testing I've done so far suggests the card is production quality in all sense of the words. I've not found any exceptions as yet. (And that's based on 21 years of experience working for a large telephone company as an engineer.) I will be getting a TDM2400 to eval as well, but it will likely take a week or two before I can get it implemented/tested. I'm going to be very interested in testing fax through these cards as there does not seem to be any reasonable pstn analog interface solutions for small asterisk installations (small defined as less then a PRI). Rich Thanks for the info Rich, I have been eagerly awaiting feedback on this new card as I have yet to find anything I would consider production quality. It sounds like this new card may have finally done that. I am also curious about the new Digium TDM2400 with echo cancel. It does not look like as good of a design mechanically, it apparently does not have as good an echo canceller (not 100% sure about that) and it's quite a bit more expensive so the Sangoma is looking quite good these days. -Original Message- FYI... Just installed one of the new Sangoma A200D analog pstn cards with the hardware echo canceller on a trial basis. The card has four fxo interfaces. Excellent audio quality, excellent echo cancelling, and excellent audio levels. The four pstn lines at this location are rather long analog loops that have rather long echo trails. I started with a pair of x100p's a couple of years ago, swapped those out for one of the first TDM04b cards, had the TDM04b replaced with a later revision (H), and have always had at least some echo on pstn calls. Our pstn lines have a -7.1 measured loss from the CO's milliwatt generator. I've configured this new card with gains of 7.0 db to compensate for that loss, and audio level is now extemely good. Presumably the Sangoma hardware canceller handles much longer echo tails, and those tails have been completely eliminated. The card's setup was not exactly clear has the documentation for this new card is somewhat fragmented across multiple readme's, etc. Based on about one hour's worth of use, I'd recommend this card over everything that I've tested. (Testing has included several ata type devices plus the x100p and tdm card.) Will be testing analog fax and many other items over the next several days/weeks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On Tuesday 21 Feb 2006 23:16, Chris Bagnall wrote: £40! That would be a cheap and nasty switch with no prospect of any management. A managed switch is worth its weight in gold, /especially/ when you have to look after things remotely. How does one justify the extra cost of a managed switch for an office of no more than 5-10 users with limited SMB file sharing and lightweight internet access going over the thing? It's just not doable. In larger organizations, I agree entirely, a managed switch *is* worth its weight in gold, but not for small businesses. You are lucky then that you have never been in a position to try and work out why a node or network does not work when you are many miles away. How much do you charge a day? The chances are that just one days callout would pay for it. Using anything else other than a managed switch for a business smacks of incompetence. It can also tell if your customers have been playing sillybuggers with the network. B -- http://www.mailtrap.org.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] did from sip trunk
I want to do inbound routing of calls comming from sip trunks. Is there a way to force the DID that comes from a trunk that does not have DID support? (something like using the outgoing caller-id for the trunk?) My problem is this: I've got several sip trunks (SPA3000). I want to have an IVR in all but one of them, the one that is connected to a cellular adapter. In this line I want to let it ring until somebody picks up because many times we checks the caller-id and calls him back. One way to do that is to put this special sip trunk in its own context, create a dialplan context specific to this trunk, and do whatever you need to do within that dialplan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoIP and Asterisk
James Harper a écrit : I want to do it the other way around. Asterisk---TDMoIPRADE1Telco You'll need 2 RAD boxes, i.e. Asterisk - RAD - TDMoIP - RAD - E1 - Telco I think you missed the point of my question, which was to know if there was any attempt to make Asterisk talk TDMoIP directly, so that I wouldn't have to have any E1 hardware in the Asterisk server at all to satisfy my failover requirements. I guess your answer is an indirect to my knowledge, it hasn't been done. The fonebridge would do what I want, but I can't get it in Australia, and if I imported one, I wouldn't legally be able to attach it to the phone system. I think the RAD offering is cheaper too. Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting disconnect on TDM400P with 3 FXO ports and 1 FXS port
Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to help Asterisk notice the disconect? This are the relevant parts of my zapata.conf: Callwaiting=no Usecallingpres=yes Callwaitingcallerid=yes Threewaycalling=no Transfer=yes Cancallforward=yes Callreturn=yes Echocancel=yes Echocancewhenbridged=no Echotraining=800 Rxgain=0.0 Txgain=0.0 Group=0 Callgroup=1 Pickupgroup=1 Faxdetect=incoming Immediate=yes Signaling=fxs_ks Context=from_rtc Busydetect=yes Channel = 4 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On Wed, 22 Feb 2006, Cory Andrews wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?tx_ttnews[tt_news]=354tx_ttnews[backPid]=33cHash=1bb97caf5cL=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?L=1 Looks like snom wants to compete with the aastra 9112i and the polycom ip301. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 601 Buddy Watch problems
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On Wed, 22 Feb 2006, Clint Sharp wrote: 2) GXP-2000: Not much better than the Budgetones, but at least the firmware [...[ that phone's quality). The speakerphone is useless due to echo issues. speakerphone echo bug was fixed in 1.0.1.12 -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Wed, 22 Feb 2006, The VoIP Connection wrote: The 941/942 are very nice phones. They are well made and so far the firmware seems very solid, but like their Cisco brethren they are a little expensive for what they offer in my opinion. If they were 25-30% cheaper I would be a lot more enthusiastic. If the 941 was priced like the 841 it would be a homerun. does the 942 have two 10meg ports or two 100meg ports? and is it poe only, or does it have the option of being powered from a wallwart without a poe injector? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with voicemail
Hello list Recently I run into some serious voicemail problems. When I call voicemailmain the prompts are presented ok if the voicemailbox is empty. If the voicemailbox contains messages the voicemail application exits with a "non-zero status" either when reading the number of messages or when selecting 1 for listening to new messages. I use Asterisk 1.2.4 on FreeBSD 5.4 without zaptel and G729a codecs installed. Any one with any pointers or idea Regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TFTP server for GrandStream BT phones / need testing
hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for debian/sarge at http://projects.hudecof.net/linux/atftp/ best regards Peter Hudec -- Linux hackers are funny people: They count the time in patchlevels. -- Martin Josefsson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Mode supported by VoiceMail Application
Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working ( cf http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode) Did anyone succeed that ? Thanks a lot ! JMS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Mode supported by VoiceMail Application
Jean-Marc Salsa [EMAIL PROTECTED] writes: Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... This depends on what your users are using. If you are using a Grandstream device you can configure in its administration interface which dtmf mode the telefone should use. If your IP phone is configured to use rfc2833 for example then you would write dtmfmode=rfc2833 in your sip.conf. If all users use the same dtmfmode it should be ok to write this to the general section. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoIP and Asterisk
I think you missed the point of my question, which was to know if there was any attempt to make Asterisk talk TDMoIP directly, so that I wouldn't have to have any E1 hardware in the Asterisk server at all to satisfy my failover requirements. I guess your answer is an indirect to my knowledge, it hasn't been done. To my knowledge, it hasn't been done. The fonebridge would do what I want, but I can't get it in Australia, and if I imported one, I wouldn't legally be able to attach it to the phone system. No, the phone bridge does TDMoE, not TDMoIP. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Mode supported by VoiceMail Application
Thanks, But, I do not have phones connected to Asterisk ... but only one peer : my softswitch ... So call flow is Phone - Softswitch - Asterisk - Voicemail Ican force the link Sofswitch - Asterisk ( Codec and DMTF Mode ) Codec is PCMx ... but as i said inband config is not working all the time ! Let me know if you think something else ... JMS On 2/22/06, Fabian Müller [EMAIL PROTECTED] wrote: Jean-Marc Salsa [EMAIL PROTECTED] writes: Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ...This depends on what your users are using. If you are using a Grandstream device you can configure in its administration interfacewhich dtmf mode the telefone should use. If your IP phone isconfigured to use rfc2833 for example then you would writedtmfmode=rfc2833 in your sip.conf. If all users use the samedtmfmode it should be ok to write this to the general section.Fabian Müller___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime queues with Firebird SQL through unixodbc
Hello, I have a partially working Realtime Queue using unixODBC and Firebird. I followed the steps on asterisk-guru and the voip-info asterisk message boards but have been having no success. I say that my queue is partially working because callers can join the queue. However the calls do not go to the agents even if the agent is not tied up and is available for calls. So the caller is stuck in the queue forever. Here is some information about my setup: Computer - Dell Poweredge 1400 (2 X P3 @ 866MHz with 1GB RAM) Linux Distro - CentOS 4.2 Kernel - 2.6.9-22.0.1.ELsmp Asterisk Version - 1.2.4 unixODBC version - 2.2.9-1 Firebird Super Server version - 1.5.3 Firebird ODBC driver version - 1.2.0.69 Here is my CLI output during the call: -- Executing Answer(IAX2/117-14, ) in new stack -- Executing Queue(IAX2/117-14, tsupport|hH) in new stack -- Started music on hold, class 'default', on channel 'IAX2/117-14' Feb 9 18:15:44 WARNING[32567]: channel.c:2535 ast_request: No channel type registered for '' -- Stopped music on hold on IAX2/117-14 -- Playing 'queue-youarenext' (language 'en') -- Registered IAX2 '123' (AUTHENTICATED) at 192.168.0.111:4569 -- Told IAX2/117-14 in tsupport their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Started music on hold, class 'default', on channel 'IAX2/117-14' -Nothing wrong here aside from getting a warning (which should not be there in the first place). I check if realtime is loading properly so I do a realtime load: realtime load queues name tsupport Column Name Column Value NAME tsupport MUSICONHOLD default ANNOUNCE technical-support TIMEOUT 5 MONITOR_JOIN 1 MONITOR_FORMAT gsm ANNOUNCE_FREQUENCY 60 ANNOUNCE_ROUND_SECONDS 0 ANNOUNCE_HOLDTIME once RETRY 2 WRAPUPTIME 0 SERVICELEVEL 30 STRATEGY random JOINEMPTY strict LEAVEWHENEMPTY no EVENTMEMBERSTATUS 1 EVENTWHENCALLED 1 REPORTHOLDTIME 1 MEMBERDELAY 2 I check my members: realtime load queue_members queue_name tsupport Column Name Column Value QUEUE_NAME tsupport INTERFACE Agent/1000 I check the queues: show queues tsupport has 0 calls (max unlimited) in 'random' strategy (0s holdtime), W:0, C:0, A:4, SL:0.0% within 30s Members: (Invalid) has taken no calls yet No Callers -this shows that I have invalid members despite having agent 1000 logged on (using AgentLogin app). I think this has something to do with how I setup the datatypes for my queus and queue member tables as I have no problems having asterisk save my CDR through unixODBC with firebird nor does my iax peers/users table have any problems either. Below is how I set up my queues_table and queue_member_table: CREATE TABLE queue_table ( name VARCHAR(128) NOT NULL PRIMARY KEY, musiconhold VARCHAR(128), announce VARCHAR(128), context VARCHAR(128), timeout NUMERIC(11,0), monitor_join SMALLINT, monitor_format VARCHAR(128), queue_youarenext VARCHAR(128), queue_thereare VARCHAR(128), queue_callswaiting VARCHAR(128), queue_holdtime VARCHAR(128), queue_minutes VARCHAR(128), queue_seconds VARCHAR(128), queue_lessthan VARCHAR(128), queue_thankyou VARCHAR(128), queue_reporthold VARCHAR(128), announce_frequency NUMERIC(11,0), announce_round_seconds NUMERIC(11,0), announce_holdtime VARCHAR(128), retry NUMERIC(11,0), wrapuptime NUMERIC(11,0), maxlen NUMERIC(11,0), servicelevel NUMERIC(11,0), strategy VARCHAR(128), joinempty VARCHAR(128), leavewhenempty VARCHAR(128), eventmemberstatus SMALLINT, eventwhencalled SMALLINT, reportholdtime SMALLINT, memberdelay NUMERIC(11,0), weight NUMERIC(11,0), timeoutrestart SMALLINT ); CREATE TABLE queue_member_table ( queue_name varchar(100) NOT NULL, interface varchar(100) NOT NULL, penalty NUMERIC(11,0), PRIMARY KEY (queue_name, interface) ); I am having problems with creating the queue_member_table. Using varchar(128) as listed on both the asteriskguru and voip-info websites results with firebird telling me a 'key size too big for index' error. setting the size to 100 eliminates this problem and creates the table properly. -- Best Regards, Eric Piros Optimum Source Inc. e-mail: [EMAIL PROTECTED] msn: [EMAIL PROTECTED] ym: ericpiros aol: ericpiros icq: 273509181 ph: +63(2)-914 ext 109 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial and Congestion
Hi folks,very stupid question, how do I setup a Dial with multiple Zap choises?I've setup this, but maybe is wrong:exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(Zap/g2/${EXTEN})exten = _7653.,3,Dial(Zap/g4/${EXTEN})exten = _7653.,101,Congestionwhat I want to do is that as soon as lines are not available on g2, all others outgoing calls must go to g4.
Should be very easy, but it do not work.If the configuration is correct, then I must check the PRI.Thanks again-- .:FaberK:.
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[Asterisk-Users] Cisco 79xx = Asterisk - SIP or SCCP?
One easy question for experienced users. Should I use Cisco VoIP phones with SIP or SCCP? What are the (dis)advantages of one or another? Please tell me your stories. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial and Congestion
On 220206, 12:54, FaberK wrote:
Hi folks,
very stupid question, how do I setup a Dial with multiple Zap choises?
I've setup this, but maybe is wrong:
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(Zap/g2/${EXTEN})
exten = _7653.,3,Dial(Zap/g4/${EXTEN})
exten = _7653.,101,Congestion
what I want to do is that as soon as lines are not available on g2, all
others outgoing calls must go to g4.
Should be very easy, but it do not work.
If the configuration is correct, then I must check the PRI.
It should be like this:
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,ChanIsAvail(Zap/g2)
exten = _7653.,3,Dial(Zap/g2/${EXTEN})
exten = _7653.,103,Dial(Zap/g4/${EXTEN})
exten = _7653.,204,playtones(congestion)
Cheers
--
Massimiliano Stucchi
WillyStudios.com
[EMAIL PROTECTED]
Http://www.willystudios.com/max/
pgpMbxvQojcBV.pgp
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[Asterisk-Users] Problem with receiving faxes with spandsp - full log included (long)
Hello, I have spandsp working with asterisk (tested on one PSTN fax machine). Today someone wanted to send me a fax, but there was a problem with the TIFF file. This is what asterisk console says: TIFFReadDirectory: /var/spool/asterisk/fax/1140608321.7.tif: cannot handle zero number of strips. MissingRequired: /var/spool/asterisk/fax/1140608321.7.tif: TIFF directory is missing required StripOffsets field. Do you have any idea what could be a problem? Below is full log from spandsp debug. Feb 22 12:38:41 VERBOSE[22100] logger.c: -- Executing RxFAX(SIP/my.cisco.router.ip-08314d98, /var/spool/asterisk/fax/1140608321.7.tif|debug) in new stack Feb 22 12:38:41 DEBUG[16200] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Match Found Feb 22 12:38:41 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:41 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:42 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:42 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Changed from phase 1 to 4 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW ???: Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Real-time Internet fax (T.38) Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW V.8 capable Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Prefer 64 octet blocks Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x90 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Supported data signalling rates: Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: V.27ter fallback mode Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW 2D coding Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Scan line length: Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: 215mm Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Recording length: Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: A4 (297mm) Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Receiver's minimum scan line time: Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: 20ms at 3.85 l/mm: T7.7 = T3.85 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x1 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Minimum scan line time for higher resolutions: T15.4 = T7.7 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Character mode Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x10 Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: FLOW DIS:Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: 80Feb 22 12:38:43 DEBUG[22100] app_rxfax.c: 00Feb 22 12:38:43 DEBUG[22100$ Feb 22 12:38:45 DEBUG[22100] app_rxfax.c: FLOW HDLC underflow in state 9 Feb 22 12:38:45 DEBUG[22100] app_rxfax.c: FLOW Changed from phase 4 to 3 Feb 22 12:38:45 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier up Feb 22 12:38:46 DEBUG[22100] app_rxfax.c: FLOW HDLC carrier down Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW T4 timeout in state 9 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Changed from phase 3 to 4 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW ???: Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Real-time Internet fax (T.38) Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW V.8 capable Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Prefer 64 octet blocks Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x90 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Supported data signalling rates: Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: V.27ter fallback mode Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW 2D coding Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Scan line length: Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: 215mm Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Recording length: Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: A4 (297mm) Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Receiver's minimum scan line time: Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: 20ms at 3.85 l/mm: T7.7 = T3.85 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x1 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Minimum scan line time for higher resolutions: T15.4 = T7.7 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Character mode Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW Reserved: 0x10 Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: FLOW DIS:Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: 80Feb 22 12:38:48 DEBUG[22100] app_rxfax.c: 00Feb 22 12:38:48 DEBUG[22100$ Feb 22 12:38:49 DEBUG[22100] app_rxfax.c: FLOW T2 timeout Feb 22 12:38:49 DEBUG[22100] app_rxfax.c: FLOW Start receiving document Feb 22 12:38:49 DEBUG[22100] app_rxfax.c: FLOW ???: Feb 22 12:38:49 DEBUG[22100] app_rxfax.c: FLOW Real-time Internet fax
[Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail on recievedvoicemail
Arne Morten Johansen wrote: As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that in Gentoo I would really appreciate it. What package provides sendmail in Gentoo? Sendmail, Postfix, Exim are all possibilities... -Barry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] did from sip trunk
2006/2/22, Rich Adamson [EMAIL PROTECTED]: One way to do that is to put this special sip trunk in its own context, create a dialplan context specific to this trunk, and do whatever you need to do within that dialplan. Yes, this must work. But it is a patch. I think it would be cleaner that if the trunk does not support did make it assumes the out caller id. I do not know much of asterisk configurations, then I couldn't find a way to do this, but I supose there must be a way. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail
It's fixed now. In /etc/mail/ssmtp.conf, this (FromLineOverride=YES) line was commented out. Removing that comment did the trick :) Now I only need to change the e-mail's title. Is that possible? -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan Sendt: 22. februar 2006 13:25 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail Arne Morten Johansen wrote: As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that in Gentoo I would really appreciate it. What package provides sendmail in Gentoo? Sendmail, Postfix, Exim are all possibilities... -Barry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail
Arne Morten Johansen wrote:
It's fixed now.
Great!
In /etc/mail/ssmtp.conf, this (FromLineOverride=YES) line was commented
out. Removing that comment did the trick :)
Now I only need to change the e-mail's title. Is that possible?
Same way. In voicemail.conf: emailsubject and emailbody
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default
shows
; just the CIDNAME, if it is not null, otherise just the CIDNUM, or an
unknown
; caller, if they are both null.
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant
;
-Barry Flanagan
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 22. februar 2006 13:25
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail
onrecievedvoicemail
Arne Morten Johansen wrote:
As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that in Gentoo I would really appreciate it.
What package provides sendmail in Gentoo? Sendmail, Postfix, Exim are
all possibilities...
-Barry
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-Barry Flanagan
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Re: [Asterisk-Users] Dial and Congestion
Hi Massimiliano,thanks for your prompt reply. Unfortunately that solution seems to not work, but I not sure is your code, I'm starting to believe that this PRI, got some problems.My system is in production, so I have to wait for more tests.
In the meantime, I thank you so much.FaberK aka Fabrizio2006/2/22, Massimiliano Stucchi [EMAIL PROTECTED]:
On 220206, 12:54, FaberK wrote: Hi folks, very stupid question, how do I setup a Dial with multiple Zap choises?
I've setup this, but maybe is wrong: exten = _7653.,1,SetCallerID(${CALLERID(number)}) exten = _7653.,2,Dial(Zap/g2/${EXTEN}) exten = _7653.,3,Dial(Zap/g4/${EXTEN})
exten = _7653.,101,Congestion what I want to do is that as soon as lines are not available on g2, all others outgoing calls must go to g4.
Should be very easy, but it do not work. If the configuration is correct, then I must check the PRI.It should be like this:exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,ChanIsAvail(Zap/g2)exten = _7653.,3,Dial(Zap/g2/${EXTEN})exten = _7653.,103,Dial(Zap/g4/${EXTEN})exten = _7653.,204,playtones(congestion)Cheers--Massimiliano Stucchi
WillyStudios.com[EMAIL PROTECTED]Http://www.willystudios.com/max/___
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SV: [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail
Thank you very much. For some reason emailsubject was not included in my
example config. Well, it's working great now.
Last question, I promise :P. Is it possible to change the date format? I want
it in Norwegian.
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 22. februar 2006 13:52
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending
e-mailonrecievedvoicemail
Arne Morten Johansen wrote:
It's fixed now.
Great!
In /etc/mail/ssmtp.conf, this (FromLineOverride=YES) line was commented
out. Removing that comment did the trick :)
Now I only need to change the e-mail's title. Is that possible?
Same way. In voicemail.conf: emailsubject and emailbody
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default
shows
; just the CIDNAME, if it is not null, otherise just the CIDNUM, or an
unknown
; caller, if they are both null.
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant
;
-Barry Flanagan
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 22. februar 2006 13:25
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail
onrecievedvoicemail
Arne Morten Johansen wrote:
As mentioned earlier I did try that. Someone suggested that there might be an
issue with sendmail not trusting the asterisk user. And the default
behaviour of that is to not allow modification of the fromstring and
serveremail. So if you have any idea how to fix that in Gentoo I would
really appreciate it.
What package provides sendmail in Gentoo? Sendmail, Postfix, Exim are
all possibilities...
-Barry
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-Barry Flanagan
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Re: [Asterisk-Users] What business IP phone to use
I like the specs on this, the only thing that it seems to be missing is POE. Anyone know if POE is going to be supported on the 300? Looks nice and I could see it for low use areas, but would suck for wall mounting if it can't do POE. On 2/22/06, Cory Andrews [EMAIL PROTECTED] wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?tx_ttnews[tt_news]=354tx_ttnews[backPid]=33cHash=1bb97caf5cL=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?L=1 Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Clint Sharp To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 22, 2006 1:03 AM Subject: Re: [Asterisk-Users] What business IP phone to use It's funny this thread has been coming up, because I've been testing out phones at my office, and I just did a fairly intensive quality test on them.1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. 2) GXP-2000: Not much better than the Budgetones, but at least the firmware is still in active development. Feature-wise it's pretty cool, but poor firmware and poor handset hardware again make this a real problem for us. We lost one handset to static electricity yesterday (which was fixed by adding in a microphone from an old business set, which actually improved that phone's quality). The speakerphone is useless due to echo issues. However, 4 line appearances is pretty cool for that price of phone, and passthrough Ethernet at 100 mbs is pretty cool too. Overall, I can't recommend them, because while they sound slightly better than the budgetones, I still get many complaints about muffled calls. 3) Polycom: Of the 4 phone brands we're actively using (not including the Wifi phone which rarely gets used), this was the best until I got the Snom in today. The handset is of good quality. I have an IP 301, but if the cheapest phone is this good, I'd definitely get a 501 or 601 (and am considering ordering some, although I may order Snom 320s instead). Their support policies do get on my nerves, I'd like to not have to worry about what reseller I'm using, but it's a solid phone with solid features, although the menus are cumbersome and I haven't gotten MWI to work on it yet. 4) Snom 320: This is an excellent phone based off one days testing. Minimal configuration, professional looking web interface, and the best sound quality of any of the phones I tested. THe speakerphone works great, and the handset quality is outstanding, and tested the best with my callers that were listening to me through the PSTN. I haven't upgraded firmware or anything on this yet, so can't tell you there, but I can't see a compelling reason to upgrade from whatever it shipped with that this point (i'm not feature crazy, I only upgrade the firmware if basic features don't seem to be working right). Overall, stay away from the Grandstream's IMHO. The audio quality issues will drive you insane. I'm hoping someone will come out with a sub-$100 phone that drops some features but fixes what should be the cheapest part of the phone to manufacture, since they've been the same for nearly 50 years, the handset. Clint ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice conferencing server capacity
Hello,We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created.The rest of the week, several small conferences will take place (5-10 simultaneous users) using 5-10 conference rooms.Is the Dell PE 2850 3GHz with 2G memory up to the task? If not, will adding another processor solve it?These entries from the dimensioning portion of the Wiki gives me hope that id does http://www.voip-info.org/wiki/view/Asterisk+dimensioning Capacity of MeetMe: With 28 persons on a Pentium II, 300MHz, 128 MB vmstat shows 70% idle. "I have anywhere from 15, to a peak max of 30 traders all using the same meetme conf during the day. My * is running on a old 4U 500Mhz machine (dual board, one processor installed now). With the exception of of a few problems from software sip phones, our implementation has been relatively problem free.I just want to get more opinions from others especially those who have built conferencing servers recently.Thank you very much.richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tormenta CAS signaling
Hi Viktor, So? You seem to have 5 channels set to fxols mode OK. What exactly that does, I am not sure. FXO operation is highly telco dependent with E1s. I believe what is implemented in zaptel is an arrangement to work with some E1 channel banks. Do you have an actual question this time? Steve Viktor Tatianin wrote: Hi Steve I have next config zttool Current Alarms: No alarms. Sync Source:Tormenta 2 (PCI) Quad E1 Card IRQ Misses: 0 Bipolar Viol:14 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 6/ 6 112333 1234567890123456789012345789012 TxA 00 TxB 11 TxC 00 TxD 11 RxA 11 RxB 01 RxC 01 RxD 11 ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: FXO Loopstart (Default) (Slaves: 32) Channel 33: FXO Loopstart (Default) (Slaves: 33) Channel 34: FXO Loopstart (Default) (Slaves: 34) Channel 35: FXO Loopstart (Default) (Slaves: 35) Channel 36: FXO Loopstart (Default) (Slaves: 36) Channel 37: FXO Loopstart (Default) (Slaves: 37) 37 channels configured. this is zaptel.conf ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-isdn-external ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks signalling=pri_cpe switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown callerid=asreceived usedistinctiveringdetection=yes usecallingpres=yes ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel= 1-15,17-31 signalling=fxo_ks context=bank group=1 cas=32-37:1101 channel= 32-37 This is my zapata.conf # Global data span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us span=2,1,0,cas,hdb3 fxols=32-37 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Tuesday, February 21, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Viktor Tatianin wrote: Hi Steve I attempt change in zapata.conf cas=1-15:1101 but use zttool view ABCD bits 1010 Regards, Viktor Have you put the E1 in CAS mode with something like: span=1,1,0,cas,hdb3 Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Friday, February 10, 2006 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Viktor Tatianin wrote: Hello Can anyone know how may change(inverting) cas signaling ABCD bits at the Tormenta 2 (four E1 ports) cards My cards send idle code ABCD 0101 but my mux which use as channel bank wait ABCD 1001 The idle code is set in
Re: [Asterisk-Users] What business IP phone to use
Having just read this thread from start to present I'd like to offer that I really like my Polycom 600/601s. the 501a are ok too. But I actually use an Aastra 480i CT personally. It's a great phone. Costs a little more but is by far the best I've used. Easy to setup. Central provisioning. Firmware issolid. Supports Asterisk. I'm s happpy to be rid of the ATA-Cordless combination. Michael --Original Message Text--- From: Joe Pukepail Date: Wed, 22 Feb 2006 07:20:17 -0600 I like the specs on this, the only thing that it seems to be missing is POE. Anyone know if POE is going to be supported on the 300? Looks nice and I could see it for low use areas, but would suck for wall mounting if it can't do POE. On 2/22/06, Cory Andrews [EMAIL PROTECTED] wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?_ttnews[tt_news]=354_ttnews[backPid]=33=1bb97caf5c=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?=1 Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Clint Sharp To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 22, 2006 1:03 AM Subject: Re: [Asterisk-Users] What business IP phone to use It's funny this thread has been coming up, because I've been testing out phones at my office, and I just did a fairly intensive quality test on them. 1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. 2) GXP-2000: Not much better than the Budgetones, but at least the firmware is still in active development. Feature-wise it's pretty cool, but poor firmware and poor handset hardware again make this a real problem for us. We lost one handset to static electricity yesterday (which was fixed by adding in a microphone from an old business set, which actually improved that phone's quality). The speakerphone is useless due to echo issues. However, 4 line appearances is pretty cool for that price of phone, and passthrough Ethernet at 100 mbs is pretty cool too. Overall, I can't recommend them, because while they sound slightly better than the budgetones, I still get many complaints about muffled calls. 3) Polycom: Of the 4 phone brands we're actively using (not including the Wifi phone which rarely gets used), this was the best until I got the Snom in today. The handset is of good quality. I have an IP 301, but if the cheapest phone is this good, I'd definitely get a 501 or 601 (and am considering ordering some, although I may order Snom 320s instead). Their support policies do get on my nerves, I'd like to not have to worry about what reseller I'm using, but it's a solid phone with solid features, although the menus are cumbersome and I haven't gotten MWI to work on it yet. 4) Snom 320: This is an excellent phone based off one days testing. Minimal configuration, professional looking web interface, and the best sound quality of any of the phones I tested. THe speakerphone works great, and the handset quality is outstanding, and tested the best with my callers that were listening to me through the PSTN. I haven't upgraded firmware or anything on this yet, so can't tell you there, but I can't see a compelling reason to upgrade from whatever it shipped with that this point (i'm not feature crazy, I only upgrade the firmware if basic features don't seem to be working right). Overall, stay away from the Grandstream's IMHO. The audio quality issues will drive you insane. I'm hoping someone will come out with a sub-$100 phone that drops some features but fixes what should be the cheapest part of the phone to manufacture, since they've been the same for nearly 50 years, the handset. Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] What business IP phone to use
The PCB has PoE "prepared" - if you open it you will see that there is a lot of space where you can solder all kinds of resistors and capacitors. Thats for PoE. However we decided that we don't place the necessary components because it would increase the price to the end customer by 25 USD - which would take us into a different pricing region. But apart from that we put everything else from the snom 320/360 there. And IMHO the audio quality is nothing less than the "high end" models, the handsfree mode probably even better (we avoided some mistakes we made in the other models). Even the 3-way conference is supported. Low use?! I would say at least 80 % of phone users today are "low use".A phone with great audio and mandatory (but not sexy) features like security for a mainstream price was missing for those users. And yes, I am from snom... (see my address!). Please excuse my excitement. Christian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe PukepailSent: Wednesday, February 22, 2006 8:31 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] What business IP phone to use I like the specs on this, the only thing that it seems to be missing is POE. Anyone know if POE is going to be supported on the 300? Looks nice and I could see it for low use areas, but would suck for wall mounting if it can't do POE. On 2/22/06, Cory Andrews [EMAIL PROTECTED] wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?tx_ttnews[tt_news]=354tx_ttnews[backPid]=33cHash=1bb97caf5cL=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?L=1 Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Clint Sharp To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 22, 2006 1:03 AM Subject: Re: [Asterisk-Users] What business IP phone to use It's funny this thread has been coming up, because I've been testing out phones at my office, and I just did a fairly intensive quality test on them.1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. 2) GXP-2000: Not much better than the Budgetones, but at least the firmware is still in active development. Feature-wise it's pretty cool, but poor firmware and poor handset hardware again make this a real problem for us. We lost one handset to static electricity yesterday (which was fixed by adding in a microphone from an old business set, which actually improved that phone's quality). The speakerphone is useless due to echo issues. However, 4 line appearances is pretty cool for that price of phone, and passthrough Ethernet at 100 mbs is pretty cool too. Overall, I can't recommend them, because while they sound slightly better than the budgetones, I still get many complaints about muffled calls. 3) Polycom: Of the 4 phone brands we're actively using (not including the Wifi phone which rarely gets used), this was the best until I got the Snom in today. The handset is of good quality. I have an IP 301, but if the cheapest phone is this good, I'd definitely get a 501 or 601 (and am considering ordering some, although I may order Snom 320s instead). Their support policies do get on my nerves, I'd like to not have to worry about what reseller I'm using, but it's a solid phone with solid features, although the menus are cumbersome and I haven't gotten MWI to work on it yet. 4) Snom 320: This is an excellent phone based off one days testing. Minimal configuration, professional looking web interface, and the best sound quality of any of the phones I tested. THe speakerphone works great, and the handset quality is outstanding, and tested the best with my callers that were listening to me through the PSTN. I haven't upgraded firmware or
Re: [Asterisk-Users] Cannot see the caller id , When calls made from one server to another
--- yusuf [EMAIL PROTECTED] wrote:
John Joseph wrote:
Hi
I had installed and configured 2 IAX server ,
users from 1'st server can dial to the second
server
and vice versa
But when I make calls to users in
other
server , on my client , I get the caller if as
[EMAIL PROTECTED] , the same I get when I
try
reverse , ie I get on my cleint caller id as
[EMAIL PROTECTED]
Please guide me what should I
do
for displaying the user id when users from one
server calls to other server users [ I get the
call-ID when users dial to the same server they
are ]
part of my iax.conf file is as follows
(20.32)
[johnb]
type=friend
user=johna
secret=secret
host=192.168.20.99
context=project
extensions.conf (20.32)
exten = _4XXX,1,Dial(IAX2/johnb/${EXTEN:1},30,r)
exten = _4XXX,2,Congestion
my iax.conf file in (20.99) is
[johna]
type=friend
user=johna
secret=secret
host=192.168.20.32
context=project
extensions.conf (20.99) contains
exten = _3XXX,1,Dial(IAX2/johna/${EXTEN:1},30,r)
exten = _3XXX,2,Congestion
John,
in iax.conf, there are settings for callerid,
so try callerid=johna
or callerid=asrecieved
and usecallerid=yes
should do it
yusuf
Thanks yusuf
it worked when i did usecallerid=yes in both the
servers , from 20.32 when I make a call receiver ,
seems the correct name
but from 20.99 when I make a call receiver sees
the caller id as [EMAIL PROTECTED]
thanks a lot
Joseph John
___
Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail
http://uk.messenger.yahoo.com
___
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RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
Hi all, Thanks for your replies and sharing your experiences. Is there any way in SPA3000 to send the rings to sip phones on asterisk while still waiting for the caller ID? This will affect the dial plan sequence but maybe user will have the option to pickup right away or wait until the caller ID displays. Or maybe there is a way for SPA3000 to find the caller ID a littler faster, as all the other phones do which are directly connected to the Bell line. Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proxy Authentication Required issue
How does Asterisk authenticate the user to make calls? The issue I am running into is that when the user sends an Invite to the asterisk which is returned by a 407, which is in turn is replied with a new Invite and the proper authentication information included in it, the Asterisk sends a second 407...and the call does not go through...The setup is as follows:X-Lite - SBC - AsteriskThe problem only occurs when I have the SBC in the middle, and from what I can see, the only difference with the SBC in place is that the second Invite which has the proper authentication in it, is sent from the SBC to the Asterisk with a NEW call-ID and Cseq # instead of using the initial INVITE. Therefore, the SBC is actually changing this information and treats it as a new call when the client does send it with the same call-ID and an incremented Cseq as the first Invite.So the question in mind is the following:Is it REQUIRED for Asterisk that thesecond Invite which is a response to the 407 that has the proper authentication information to use the same call ID and Cseq or this should not matter?thanks, Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
Do you know when it's coming out? What will the price be?- WaldoOn Feb 22, 2006, at 1:18 AM, Cory Andrews wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?tx_ttnews[tt_news]=354tx_ttnews[backPid]=33cHash=1bb97caf5cL=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?L=1 Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
Thanks for your replies and sharing your experiences. Is there any way in SPA3000 to send the rings to sip phones on asterisk while still waiting for the caller ID? This will affect the dial plan sequence but maybe user will have the option to pickup right away or wait until the caller ID displays. Or maybe there is a way for SPA3000 to find the caller ID a littler faster, as all the other phones do which are directly connected to the Bell line. No, there is no way to do that in the spa3k. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
From: Christian Stredicke [EMAIL PROTECTED] The PCB has PoE prepared - if you open it you will see that there is a lot of space where you can solder all kinds of resistors and capacitors. Thats for PoE. However we decided that we don't place the necessary components because it would increase the price to the end customer by 25 USD - which would take us into a different pricing region. But apart from that we put everything else from the snom 320/360 there. And IMHO the audio quality is nothing less than the high end models, the handsfree mode probably even better (we avoided some mistakes we made in the other models). Even the 3-way conference is supported. Low use?! I would say at least 80 % of phone users today are low use. A phone with great audio and mandatory (but not sexy) features like security for a mainstream price was missing for those users. And yes, I am from snom... (see my address!). Please excuse my excitement. What is the expected target date for efforts to begin filling the reseller channel? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx firmware
Hi, as far as I know, you can't actually buy the firmware, you have to get the service contract, I talked to the guy at CDW who talked to his Cisco guy, and they told me to buy a $92 service contract. hope that helps.. On Wed, 2006-02-22 at 10:34 +0100, Tomislav Parčina wrote: I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me technical support agreement which cost around 75$ for every phone! -- Tomislav Parcina tparcina#lama.hr -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best ATA for general residential deployment??
I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want to set it up and never look at it again ;) The way you make money on lower cost residential services is to make sure you spend as little labor as possible after the fact. If we have to install a $200 part, we'll make that money back with the monthly fee over time as long as we don't have to go back to, it or replace it, or spend a lot of time on the phone doing support. 2) Second priority is remote provisioninga truck roll to change configurations is not acceptable. A web or telnet interface is tolerable, but tftp or http auto configuration is desireable. 3) Third priority is pricefor obvious reasons Perhaps the biggest issue is we don't want to have to supply a router or switch in addition to the ATA. It's a lot of extra cabling that people might screw up, extra parts that might break, extra time for the installation, etc. Ideally, either a device that functions as an ethernet bridge (like vonage ATA's) so that it can be positioned in-line with other equipment; or a combination router/SIP adapter. The absolute best thing in the world might be a combination router, 802.11 AP, 4 port ethernet switch, and SIP adapter with a backup battery. Plug in one box and you're done. If the router can be reconfigured as a bridge (for customers who prefer their own router) so much the better. Any reccomendations would be welcome. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with voicemail
Roger Lewau wrote: If the voicemailbox contains messages the voicemail application exits with a non-zero status either when reading the number of messages or when selecting 1 for listening to new messages. Is it possible the permissions for the sounds directory or individual files within have changed such that the user asterisk runs under no longer has read access? -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tormenta CAS signaling
Hi Steve May I change TX ABCD bits 0101 to 1001 or no ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Wednesday, February 22, 2006 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Hi Viktor, So? You seem to have 5 channels set to fxols mode OK. What exactly that does, I am not sure. FXO operation is highly telco dependent with E1s. I believe what is implemented in zaptel is an arrangement to work with some E1 channel banks. Do you have an actual question this time? Steve Viktor Tatianin wrote: Hi Steve I have next config zttool Current Alarms: No alarms. Sync Source:Tormenta 2 (PCI) Quad E1 Card IRQ Misses: 0 Bipolar Viol:14 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 6/ 6 112333 1234567890123456789012345789012 TxA 00 TxB 11 TxC 00 TxD 11 RxA 11 RxB 01 RxC 01 RxD 11 ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: FXO Loopstart (Default) (Slaves: 32) Channel 33: FXO Loopstart (Default) (Slaves: 33) Channel 34: FXO Loopstart (Default) (Slaves: 34) Channel 35: FXO Loopstart (Default) (Slaves: 35) Channel 36: FXO Loopstart (Default) (Slaves: 36) Channel 37: FXO Loopstart (Default) (Slaves: 37) 37 channels configured. this is zaptel.conf ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-isdn-external ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks signalling=pri_cpe switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown callerid=asreceived usedistinctiveringdetection=yes usecallingpres=yes ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel= 1-15,17-31 signalling=fxo_ks context=bank group=1 cas=32-37:1101 channel= 32-37 This is my zapata.conf # Global data span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us span=2,1,0,cas,hdb3 fxols=32-37 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Tuesday, February 21, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Viktor Tatianin wrote: Hi Steve I attempt change in zapata.conf cas=1-15:1101 but use zttool view ABCD bits 1010 Regards, Viktor Have you put the E1 in CAS mode with something like: span=1,1,0,cas,hdb3 Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Friday, February 10, 2006 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:
Re: [Asterisk-Users] Best ATA for general residential deployment??
Try the Sipura SPA-2002.. at good prices from VoipSupply.com We have been using those now with 0 problems. We remote provision them from our office here. Once a minute (time configurable) each device checks in with us to check out its configuration file and see if it needs updates. The devices run around $60 a piece, so they are pretty cheap as well. On 2/22/06, Adam Moffett [EMAIL PROTECTED] wrote: I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want to set it up and never look at it again ;) The way you make money on lower cost residential services is to make sure you spend as little labor as possible after the fact. If we have to install a $200 part, we'll make that money back with the monthly fee over time as long as we don't have to go back to, it or replace it, or spend a lot of time on the phone doing support. 2) Second priority is remote provisioninga truck roll to change configurations is not acceptable. A web or telnet interface is tolerable, but tftp or http auto configuration is desireable. 3) Third priority is pricefor obvious reasons Perhaps the biggest issue is we don't want to have to supply a router or switch in addition to the ATA. It's a lot of extra cabling that people might screw up, extra parts that might break, extra time for the installation, etc. Ideally, either a device that functions as an ethernet bridge (like vonage ATA's) so that it can be positioned in-line with other equipment; or a combination router/SIP adapter. The absolute best thing in the world might be a combination router, 802.11 AP, 4 port ethernet switch, and SIP adapter with a backup battery. Plug in one box and you're done. If the router can be reconfigured as a bridge (for customers who prefer their own router) so much the better. Any reccomendations would be welcome. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice conferencing server capacity
On 2/22/06, Richard OSS [EMAIL PROTECTED] wrote: Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week, several small conferences will take place (5-10 simultaneous users) using 5-10 conference rooms. Is the Dell PE 2850 3GHz with 2G memory up to the task? If not, will adding another processor solve it? These entries from the dimensioning portion of the Wiki gives me hope that id does http://www.voip-info.org/wiki/view/Asterisk+dimensioning Capacity of MeetMe: With 28 persons on a Pentium II, 300MHz, 128 MB vmstat shows 70% idle. I have anywhere from 15, to a peak max of 30 traders all using the same meetme conf during the day. My * is running on a old 4U 500Mhz machine (dual board, one processor installed now). With the exception of of a few problems from software sip phones, our implementation has been relatively problem free. I just want to get more opinions from others especially those who have built conferencing servers recently. Speaking from experience, I'd expect that this setup is more than adequate. You may want to disable hyperthreading though on the procs if you can. There really isn't an advantage to having it enabled as the additional CPU has no FPU. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Problems with voicemail
Hello At the moment we are running asterisk as root. But I just checked and did chmod 777 to be certain, but the problem still remains. This is the output from asterisk. Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' Regards Roger -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Johnathan Corgan Skickat: den 22 februari 2006 15:52 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Problems with voicemail Roger Lewau wrote: If the voicemailbox contains messages the voicemail application exits with a non-zero status either when reading the number of messages or when selecting 1 for listening to new messages. Is it possible the permissions for the sounds directory or individual files within have changed such that the user asterisk runs under no longer has read access? -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring
If not in spa3k, then how about digium hardware, will that be faster in picking up caller IDs or is it possible to make it work faster. I need only one FXS/FXO. Is X101P single FXS/FXO? Zach A. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring Thanks for your replies and sharing your experiences. Is there any way in SPA3000 to send the rings to sip phones on asterisk while still waiting for the caller ID? This will affect the dial plan sequence but maybe user will have the option to pickup right away or wait until the caller ID displays. Or maybe there is a way for SPA3000 to find the caller ID a littler faster, as all the other phones do which are directly connected to the Bell line. No, there is no way to do that in the spa3k. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Follow Me
Thank You. On 2/21/06, C F [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. On 2/21/06, Max Clark [EMAIL PROTECTED] wrote: Hi all, I am interested in a follow me script for Asterisk - specifically I am looking for one that will prompt the calling party to record their name and then call through a list of numbers playing the recording. If a digit is pressed by the recipient then the call is put through. Is there anything like this available as an example for Asterisk? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Don't have an SPA-942 here right now, but a D-Link switch detects the SPA-941 as 10base-T/half-duplex. Just like real Cisco phones, the 942 can be powered with a wall wart but it does not come with one (extra charge). -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 5:30 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: 'mustardman29' Subject: RE: [Asterisk-Users] What business IP phone to use On Wed, 22 Feb 2006, The VoIP Connection wrote: The 941/942 are very nice phones. They are well made and so far the firmware seems very solid, but like their Cisco brethren they are a little expensive for what they offer in my opinion. If they were 25-30% cheaper I would be a lot more enthusiastic. If the 941 was priced like the 841 it would be a homerun. does the 942 have two 10meg ports or two 100meg ports? and is it poe only, or does it have the option of being powered from a wallwart without a poe injector? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with SU100
Hi all, I am trying to add a TigerJet usb adapter to my asterisk installation I have the 1.2.4 asterisk bristuffed version, holding two zaphfc (ISDN) cards If I connect the TigerJet adapter to my linux box (Suse 10) i see: Bus 005 Device 001: ID : Bus 004 Device 003: ID 06e6:831c Tiger Jet Network, Inc. Bus 004 Device 001: ID : Bus 003 Device 001: ID : Bus 002 Device 001: ID : Bus 001 Device 001: ID : and in warning log: Feb 22 15:46:12 asterisk01 kernel: zaptel: module not supported by Novell, setting U taint flag. Feb 22 15:46:12 asterisk01 kernel: wcusb: module not supported by Novell, setting U taint flag. Feb 22 15:46:12 asterisk01 kernel: Wildcard USB FXS Interface driver registered So everything seems to be OK. Then I modify to the /etc/zaptel.conf file: # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 # fxsks=7 fxoks=7 according to http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+config+zaptel.confdiff=13 I think that I should consider this card as a Wildcard S100U (hence the fxoks line) but if I type ztcfg -vvv i see : SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Slaves: 07) 7 channels configured. Changing signalling on channel 7 from Clear channel to FXO Kewlstart ZT_CHANCONFIG failed on channel 7: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? nothing changes if I modify fxoks=7 to fxsks=7. What am I doing wrong ? does exist any incompatibilty between bristuff and S100U ? Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx firmware
CDW and other large resellers like them have a difficult time selling service contracts. The issue is they _must_ provide Cisco with a serial number (of the phone) which is checked by Cisco to see if the company wanting the contract was the original phone purchaser. If the phone is a used phone (from any number of used equipment resellers), they won't write the service contract unless the phone is recertified, etc. CDW is then caught in the crossfire where they really don't have the people resources to address the recertification and contract sales effort. Cisco still wants CDW (and other new equipment resellers) to sell contracts, but basically those high-volume resellers are limited to contracts on new equipment only. Of coarse, the entry-level sales telemarketing people hired by firms like CDW aren't trained or even knowledgable on Cisco's license terms, etc, and will frequently take orders on used equipment that they cannot complete. Been there, tried it, and have been around the Cisco block for 20+ years. Essentially it all boils down to Cisco trying to push new equipment verses used/recertified equipment and trying to limit the used equipment market. If one actually looks at the cost of buying a new 7960 with a license (as an example), verses buying used and having to jump through all the road blocks, buying new with a license is cheaper (under $300 US now). As a side note, there are some used equipment resellers that do have the connections to say a phone is recertified and will drop the phone onto an existing customer account, but the majority won't mess with it given the profit margins, etc. For all practical purposes, there is absolutely nothing that a reseller can do in the recertification process for a 79x0 phone other then paperwork; there is not even a way for these folks to remove the software that was installed on a used phone. The _only_ thing they can do is prove the phone is in workable condition and reinstall some cisco-perscribed firmware. Note: I don't work for Cisco or any of their resellers. Hi, as far as I know, you can't actually buy the firmware, you have to get the service contract, I talked to the guy at CDW who talked to his Cisco guy, and they told me to buy a $92 service contract. hope that helps.. On Wed, 2006-02-22 at 10:34 +0100, Tomislav Parèina wrote: I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me technical support agreement which cost around 75$ for every phone! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debugging asterisk configuration
I'm trying to create a new contex for incomming calls from certain
trunks. My problem is this calls are not checked through ext-did (for
incoming routing). The calls from standard trunks are filtered
correctly but these ones are not. Is there some way to debug what
file/line is being executed by asterisk? My custom context is this:
[from-pstn-nofax]
include = from-pstn-custominclude customizations
include = ext-did
exten = _.,1,Goto(s,1)
exten = s,1,SetVar(INCOMING=GRP-1)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours-nofax,s,1:2)
exten = s,3,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours-nofax,s,1:3)
exten =
s,4,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours-nofax,s,1:2)
exten = s,5,Goto(from-pstn-afthours-nofax,s,1)
ext-did context is this (made by amp):
[ext-did]
include = ext-did-custom
exten = s/7045,1,SetVar(FROM_DID=s/7045)
exten = s/7045,2,Goto(ext-local,211,1)
exten = s/987073366,1,SetVar(FROM_DID=s/987073366)
exten = s/987073366,2,Goto(ext-local,211,1)
exten = _X./7045,1,Goto(s/7045)
exten = _X./987073366,1,Goto(s/987073366)
--
Alejandro Vargas
___
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Re: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring
There really is no way to completely eliminate the lag, even if you disable callerid. Another workaround would be to connect a loud phone directly to your pstn line. When you hear it ring, jump up and grab your regular phone. It'll start ringing by the third ring. You'll have callerid pop up between the third and fourth ring (the 3-4 ring that the caller hears, it'll be between the 1-2 ring on your regular line). I discovered this workaround when I left a fax machine (which had a very loud ring) directly connected to my pstn line. I was surprised at how many calls came in and never got past the voice greeting (all were unfamiliar numbers, probably wrong numbers or what-not). I'm afraid for your situation, there's no way to do what you want without some kind of workaround, especially since you need callerid information. I suppose you could disable callerid detection in asterisk, and get callerid delivered to you another way (on your TV if you have a satellite receiver that supports callerid, a device that reads off the callerid to you, one of those nifty globes I've seen at radio shack, etc.). You'll still have a little bit of delay until asterisk rings your extensions, but it'll be more like 1-2 rings instead of 3. I honestly think a voice recording being played just before it rings your extensions isn't as bad an idea as you think. I use one for my residential line in addition to my business line. Haven't heard a single complaint yet. In fact I've gotten a few nice comments from it (I can customize the recording used based on callerid, leaving nice cute messages for family/friends, and the default recording for everyone else). Hope you find a solution that suits your needs. Joseph Tanner On 2/22/06, Zach A [EMAIL PROTECTED] wrote: If not in spa3k, then how about digium hardware, will that be faster in picking up caller IDs or is it possible to make it work faster. I need only one FXS/FXO. Is X101P single FXS/FXO? Zach A. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring Thanks for your replies and sharing your experiences. Is there any way in SPA3000 to send the rings to sip phones on asterisk while still waiting for the caller ID? This will affect the dial plan sequence but maybe user will have the option to pickup right away or wait until the caller ID displays. Or maybe there is a way for SPA3000 to find the caller ID a littler faster, as all the other phones do which are directly connected to the Bell line. No, there is no way to do that in the spa3k. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
True, but managed switches fail too. My suggestion, buy two cheap ones, and keep one in the box... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Tuesday, February 21, 2006 5:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use How does one justify the extra cost of a managed switch for an office of no more than 5-10 users with limited SMB file sharing and lightweight internet access going over the thing? It's just not doable. In larger organizations, I agree entirely, a managed switch *is* worth its weight in gold, but not for small businesses. Simple formula: 1. Total Revenue 2. % of revenue derived from phone usage 3. =Cost of downtime by using SoHo or consumer gear. It's not a question of if a SoHo or low cost device will screw up, it is a question of when. This is 23 years of experience talking. Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 employees, so we fall into the SMB segment. Sometimes I'm appalled by statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in your home office, or whatever, but in any kind of meaningful business context, you *always* buy the best, and you only cry once. If you argue that your business can't support that kind of cost (which is really, actually quite cheap. Anyone remember $6000 switches? I do.) then perhaps you may want to re-evaluate whether it's appropriate to use VoIP in your business in the first place. Sure, a managed switch is not a silver bullet - but it is part of a quality implementation that *is* a silver bullet. Weakest link, and all that. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Call queue design issues and suggestions
I haven't done anything with call queues specifically.I do own a copy of Developing Cisco IP Phone Services ( http://www.amazon.ca/exec/obidos/ASIN/1587050609/qid%3D1140624389/701-7618472-4656313), though I only have ever used about 45 pages of it. I also use http://search.cpan.org/~mrpalmer/Cisco-IPPhone-0.05/IPPhone.pm to make my life a little easier.On 2/22/06, Tomislav Parčina [EMAIL PROTECTED] wrote:In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know if this works for you, but I use the following mechanism. I don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff. For each queue, dialing the extension (), puts the caller into the queue (ie, a customer calling for reservations). I use ** to sign a phone into the queue and * to sign out of a queue.Good idea, maybe sometimes I'll need it. You can use the manager to see who is currently logged into a port. It doesn't take much to write a cgi script that outputs the Cisco XML for the phones. I've built a few apps that do interesting things. It would be quite easy to write an app that:It could be easy for someone with experience, but if you have never done it before (like me) it isn't like that. Can you send us what you have done? --Tomislav Parcina[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk nagios plugin
Hi,
I am using nagios to monitor asterisk via nrpe plugin
and it is working fine. But I want also to monitor the
status of the pri. I have found a plugin which uses
the asterisk manager configuration. But it doesnt
return the result of the command being executed. As
much as I want to include it, I can since I am just a
simple administrator. :)
Hope some can help me.
#!/usr/bin/perl -w
use strict;
use IO::Socket;
use Getopt::Long;
my (
$host, $username, $password, $verbose, $help,
$command,
$version,$response,$message,
$sock,
$readloop,
$s
);
my $port = 5038;
my $exitcode = 0;
sub warning {
$s = shift;
print WARNING: $s\n;
exit(1);
}
sub error {
$s = shift;
print ERROR: $s\n;
exit(2);
}
sub syntax {
$s = shift;
unless ($s =~ /Help:/) {
$s = Error: (.$s.) or $s = 'Unknown';
}
print $s\n unless ($help);
print Syntax: $0 -h host -u username -p
password [-cwv]\n;
print * --username -u Username\n;
print * --password -p Password\n;
print * --host -h Host\n;
print --port -P Port (if not using $port)\n;
print --command -c Custom command (instead of
Status)\n;
print --verbose -v Verbose\n;
print --help -h This help\n;
exit(3);
}
Getopt::Long::Configure('bundling');
GetOptions
(p=s = \$password, password=s = \$password,
u=s = \$username, username=s = \$username,
h=s = \$host, host=s = \$host,
P=s = \$port, port=s = \$port,
c=s = \$command, command=s = \$command,
H = \$help, help = \$help,
v = \$verbose, verbose= \$verbose);
syntax(Help:) if ($help);
syntax(Missing username) unless
(defined($username));
syntax(Missing password) unless
(defined($password));
syntax(Missing host) unless (defined($host));
unless ($sock = IO::Socket::INET-new(PeerAddr =
$host, PeerPort = $port, Proto = 'tcp')) {
print(Could not connect to asterisk server
.$host.:.$port.\n);
exit(2);
}
$version = $sock;
print $version if ($verbose);
print $sock Action: Login\r\nUsername:
$username\r\nSecret: $password\r\n\r\n;
print Action: Login\r\nUsername: $username\r\nSecret:
$password\r\n\r\n if ($verbose);
$response = $sock; $message = $sock; $s = $sock;
print $response.$message if ($verbose);
print $s if ($verbose);
if ($response =~ /Response:\s+(.*)[\r\n]/) {
$response = $1;
unless ($response =~ /Success/) {
exit(1);
}
}
print $sock Action: Logoff\r\n\r\n;
print Nagios responded ok.;
0
Regards,
Leonimar
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Re: [Asterisk-Users] Tormenta CAS signaling
Viktor Tatianin wrote: Hi Steve May I change TX ABCD bits 0101 to 1001 or no ? Not when you use fxols. You can only control them when you use cas as the signalling type. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Wednesday, February 22, 2006 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Hi Viktor, So? You seem to have 5 channels set to fxols mode OK. What exactly that does, I am not sure. FXO operation is highly telco dependent with E1s. I believe what is implemented in zaptel is an arrangement to work with some E1 channel banks. Do you have an actual question this time? Steve Viktor Tatianin wrote: Hi Steve I have next config zttool Current Alarms: No alarms. Sync Source:Tormenta 2 (PCI) Quad E1 Card IRQ Misses: 0 Bipolar Viol:14 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 6/ 6 112333 1234567890123456789012345789012 TxA 00 TxB 11 TxC 00 TxD 11 RxA 11 RxB 01 RxC 01 RxD 11 ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: FXO Loopstart (Default) (Slaves: 32) Channel 33: FXO Loopstart (Default) (Slaves: 33) Channel 34: FXO Loopstart (Default) (Slaves: 34) Channel 35: FXO Loopstart (Default) (Slaves: 35) Channel 36: FXO Loopstart (Default) (Slaves: 36) Channel 37: FXO Loopstart (Default) (Slaves: 37) 37 channels configured. this is zaptel.conf ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-isdn-external ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks signalling=pri_cpe switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown callerid=asreceived usedistinctiveringdetection=yes usecallingpres=yes ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel= 1-15,17-31 signalling=fxo_ks context=bank group=1 cas=32-37:1101 channel= 32-37 This is my zapata.conf # Global data span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us span=2,1,0,cas,hdb3 fxols=32-37 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Tuesday, February 21, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Tormenta CAS signaling Viktor Tatianin wrote: Hi Steve I attempt change in zapata.conf cas=1-15:1101 but use zttool view ABCD bits 1010 Regards, Viktor Have you put the E1 in CAS mode with something like: span=1,1,0,cas,hdb3 Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of
[Asterisk-Users] Streaming Music On Hold
Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten = 1234,1,Answer exten = 1234,2,SetMusiconHold(stream2) exten = 1234,3,WaitmusiconHold(60) exten = 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI -- Executing Answer(SIP/3250072-ed28, ) in new stack -- Executing SetMusicOnHold(SIP/3250072-ed28, stream2) in new stack -- Executing WaitMusicOnHold(SIP/3250072-ed28, 60) in new stack -- Started music on hold, class 'stream2', on channel 'SIP/3250072-ed28' -- Stopped music on hold on SIP/3250072-ed28 If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I get the default music on hold. Running ngrep on port 80 shows me that the Asterisk system is not sending or receiving ANY data on port 80. What am I doing wrong? Yes, it has network and DNS connectivity. Can't believe it's this hard! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip channel status - how?
Thank you very much for that hint! I am using asterisk-java at the moment to retrieve the channel information and I now have a way of retrieving such channel information sending a sip show channels command via the manager interface. I then parse the answer from the server. But it seems that show channels consise is an even better way. I have tried it and it gives me even better information about active channels. I'll investigate the documentation of that command. Thank you so much for that excellent hint. God bless Peter It errors when you ask it for the channel 'test-1' because the parameter is the channel name, not the peer name. I've used 'show channels concise' and then parsed the output in the past. Peter Hoppe wrote: Hello! I have an asterisk setup where several sip devices are connected to an asterisk box. I am looking for a method that lets me know whether any of the sip devices is on hook / off hook / busy etc. I have tried the AGI command CHANNEL STATUS channel name but it returns with a message 'There is no channel that matches channel name' In concrete terms, my channel is a Grandstream BT 100, and I have configured it as user 'test-1'. But when I query its state I get 'There is no channel that matches test-1' When I use test-1 my log shows a device 'test-1-4-digit-hex-code', and I suppose that I need the 4-digit-hex code to enable asterisk to find the matching channel with the AGI CHANNEL STATUS command. Unfortunately that number seems to be assigned to a different value whenever the 'test-1' device is used. Is there any agi (or other fast-agi/eagi/manager/dialplan) command or technique through which I can find the state of my channel 'test-1' without me having to specify the 4-digit-hex-code? Thank you very much! Peter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * Previous message: [Asterisk-Users] sip channel status - how? * Next message: [Asterisk-Users] ztdummy on gentoo 2005.1 * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] More information about the Asterisk-Users mailing list ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel, the bug tracker or on the mailing list! /oej ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the release version. The release version is distributed as .tar.gz archives on several servers. The current released version of Asterisk is 1.2.4. The release version is fixed, we are adding no new functions and only changes it when bugs are fixed. The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. The development version is branded 1.3 and will be the basis for the next release version, version 1.4. There are also a lot of development branches in our subversion repository, hosting new functionality developed for testing by you, the asterisk community. For more information about these, please visit http://www.voip-forum.com/index.php?p=189more=1 ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. And please do not send out test messages to the list. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org You can download their new book from the web site or buy it from the bookstore. * Asterisk Daily news is at http://www.sineapps.com/news.php * VoIP-search (Asterisk mailing list etc) http://search.voip-forum.com Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. Do not use this list as a secondary support line if you do not get an answer on the -users list. It is meant for developer discussions, not advanced support. If you need answers, there is a better chance that you will get help on the irc channel. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services (asterisk-biz). You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. If you are unsure which list to use, send only to the -users list. Make sure that you remove unnecessary text when you reply, to make it easy to browse the mailing list quickly. And please do not send HTML mail to a mailing list. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new
R: [Asterisk-Users] queue behaviour
That's exactly what I was looking for. By the way, I discovered Local channels to fork into dialplan. I also discovered that roundrobin policy does not work as I expected, but that's another story. Thanks for help, _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Chris Bagnall Inviato: lunedì 20 febbraio 2006 20.21 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] queue behaviour What I'm trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I've no idea if this'll work in practice, but the theory seems sound: 1) Create some extensions in your dialplan which dial the numbers you want the queue to try: exten = 1000,1,Dial(dialstring here) exten = 1001,1,Dial(second dialstring here) etc. 2) Assign members to your queue as follows: member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] etc. 3) Set the queue to ringall or round robin as required. 4) let the list know whether it worked or not :-) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming Music On Hold
Try this: musiconhold.conf: [stream2] mode=mp3 directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr extensions.conf: exten = 1234,1,Answer exten = 1234,2,MusicOnHold(stream2) exten = 1234,3,Hangup On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote: Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten = 1234,1,Answer exten = 1234,2,SetMusiconHold(stream2) exten = 1234,3,WaitmusiconHold(60) exten = 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI -- Executing Answer(SIP/3250072-ed28, ) in new stack -- Executing SetMusicOnHold(SIP/3250072-ed28, stream2) in new stack -- Executing WaitMusicOnHold(SIP/3250072-ed28, 60) in new stack -- Started music on hold, class 'stream2', on channel 'SIP/3250072-ed28' -- Stopped music on hold on SIP/3250072-ed28 If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I get the default music on hold. Running ngrep on port 80 shows me that the Asterisk system is not sending or receiving ANY data on port 80. What am I doing wrong? Yes, it has network and DNS connectivity. Can't believe it's this hard! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fromuser required but overrides SetCallerID
Hi all, I have an asterisk box connecting to a SER instance for outbound (termination) calling. In order to authenticate with SER it seems that I have to use fromuser in the sip.conf in the peer section for the SER connection - with fromuser set I can make calls, without it I get a Forbidden - wrong password on authentication for INVITE error. The problem is that setting fromuser in the sip.conf overrides anything that I have set in the dialplan with SetCallerID. How do I work around this? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
On 2/22/06, Matt [EMAIL PROTECTED] wrote: Try the Sipura SPA-2002.. at good prices from VoipSupply.com We have been using those now with 0 problems. We remote provision them from our office here. Once a minute (time configurable) each device checks in with us to check out its configuration file and see if it needs updates. The devices run around $60 a piece, so they are pretty cheap as well. RE the remote provisioning, did you have to pay some sort of license fee to get access to the tools to generate the remote provisioning configurations and instructions on setting it all up? I have 12 x PAP2-NA/SPA-2002 and changing one setting means going around and changing all the settings for each line of the 12 ATAs, that's 24 configuration changes in total - a real PITA. If you know of a way to obtain the tools to do the remote provisioning, I'd be grateful! Thanks, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proxy Authentication Required issue
Tim Chase wrote: Is it REQUIRED for Asterisk that thesecond Invite which is a response to the 407 that has the proper authentication information to use the same call ID and Cseq or this should not matter? This is not an Asterisk requirement, it's a SIP RFC compliance requirement. The random value (nonce) that is sent as part of the initial 407 response is specific to that Call-ID, as the RFC mandates. If Asterisk receives an INVITE for a different Call-ID, it will generate a new nonce, thus ignoring the authentication info including in the INVITE. In other words: your SBC is broken. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fromuser required but overrides SetCallerID
Max Clark wrote: Hi all, I have an asterisk box connecting to a SER instance for outbound (termination) calling. In order to authenticate with SER it seems that I have to use fromuser in the sip.conf in the peer section for the SER connection - with fromuser set I can make calls, without it I get a Forbidden - wrong password on authentication for INVITE error. The problem is that setting fromuser in the sip.conf overrides anything that I have set in the dialplan with SetCallerID. How do I work around this? You convince the operator of the SIP proxy to accept your caller IDs and only authenticate on the digest auth user. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FC4 and yum install; how to configure questions
I installed FC4, ran command, # yum install asterisk. A bunch of stuff happened, but can't locate .conf files. I have a list of files: /usr/share/doc/asterisk-1.2.4/configs/features.conf.sample /usr/share/doc/asterisk-1.2.4/configs/rtp.conf.sample /usr/share/doc/asterisk-1.2.4/configs/extensions.conf.sample /usr/share/doc/asterisk-1.2.4/configs/logger.conf.sample /usr/share/doc/asterisk-1.2.4/configs/iax.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr_custom.conf.sample /usr/share/doc/asterisk-1.2.4/configs/modules.conf.sample /usr/share/doc/asterisk-1.2.4/configs/enum.conf.sample /usr/share/doc/asterisk-1.2.4/configs/oss.conf.sample /usr/share/doc/asterisk-1.2.4/configs/musiconhold.conf.sample /usr/share/doc/asterisk-1.2.4/configs/extensions.ael.sample /usr/share/doc/asterisk-1.2.4/configs/adsi.conf.sample /usr/share/doc/asterisk-1.2.4/configs/alarmreceiver.conf.sample /usr/share/doc/asterisk-1.2.4/configs/iaxprov.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr_odbc.conf.sample /usr/share/doc/asterisk-1.2.4/configs/indications.conf.sample /usr/share/doc/asterisk-1.2.4/configs/misdn.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr_manager.conf.sample /usr/share/doc/asterisk-1.2.4/configs/voicemail.conf.sample /usr/share/doc/asterisk-1.2.4/configs/dundi.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr_tds.conf.sample /usr/share/doc/asterisk-1.2.4/configs/telcordia-1.adsi /usr/share/doc/asterisk-1.2.4/configs/asterisk.adsi /usr/share/doc/asterisk-1.2.4/configs/meetme.conf.sample /usr/share/doc/asterisk-1.2.4/configs/mgcp.conf.sample /usr/share/doc/asterisk-1.2.4/configs/sip.conf.sample /usr/share/doc/asterisk-1.2.4/configs/alsa.conf.sample /usr/share/doc/asterisk-1.2.4/configs/sip_notify.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr.conf.sample /usr/share/doc/asterisk-1.2.4/configs/rpt.conf.sample /usr/share/doc/asterisk-1.2.4/configs/manager.conf.sample /usr/share/doc/asterisk-1.2.4/configs/phone.conf.sample /usr/share/doc/asterisk-1.2.4/configs/modem.conf.sample /usr/share/doc/asterisk-1.2.4/configs/codecs.conf.sample /usr/share/doc/asterisk-1.2.4/configs/dnsmgr.conf.sample /usr/share/doc/asterisk-1.2.4/configs/extconfig.conf.sample /usr/share/doc/asterisk-1.2.4/configs/adtranvofr.conf.sample /usr/share/doc/asterisk-1.2.4/configs/vpb.conf.sample /usr/share/doc/asterisk-1.2.4/configs/res_odbc.conf.sample /usr/share/doc/asterisk-1.2.4/configs/osp.conf.sample /usr/share/doc/asterisk-1.2.4/configs/queues.conf.sample /usr/share/doc/asterisk-1.2.4/configs/skinny.conf.sample /usr/share/doc/asterisk-1.2.4/configs/zapata.conf.sample /usr/share/doc/asterisk-1.2.4/configs/cdr_pgsql.conf.sample /usr/share/doc/asterisk-1.2.4/configs/privacy.conf.sample /usr/share/doc/asterisk-1.2.4/configs/festival.conf.sample /usr/share/doc/asterisk-1.2.4/configs/agents.conf.sample - - - This setup allowed me to run the command: # /usr/sbin/asterisk -c A bunch of stuff happens, then the prompt returns to the root command. I'm reading the asterisk book (just started Ch 4), and when I go looking for a .conf file, it doesn't exist. Any help appreciated. Tom -- 94% of returning troops suffer from trauma Open Studios http://www.ibiblio.org/studioforrecording/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
There is sample XML from Sipura for the 841 and 941 Phones. If you use that as a guide, you should be able to generate XML for any of the ATA boxes and then you just need to set up a server for the boxes to get their configs from. We have been doing it with pretty good success for the 841, 941, 2100 and 2002s. Darrell S. Long BestWeb Corporation Gonzalo Servat wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Try the Sipura SPA-2002.. at good prices from VoipSupply.com We have been using those now with 0 problems. We remote provision them from our office here. Once a minute (time configurable) each device checks in with us to check out its configuration file and see if it needs updates. The devices run around $60 a piece, so they are pretty cheap as well. RE the remote provisioning, did you have to pay some sort of license fee to get access to the tools to generate the remote provisioning configurations and instructions on setting it all up? I have 12 x PAP2-NA/SPA-2002 and changing one setting means going around and changing all the settings for each line of the 12 ATAs, that's 24 configuration changes in total - a real PITA. If you know of a way to obtain the tools to do the remote provisioning, I'd be grateful! Thanks, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
At 07:11 AM 02/22/2006, you wrote: If not in spa3k, then how about digium hardware, will that be faster in picking up caller IDs or is it possible to make it work faster. I need only one FXS/FXO. Is X101P single FXS/FXO? I have a TDM04 and it seems to ring about 1 ring behind. The analog phone rings, I grab the headset and about the second ring I pick up the * phone. The analog phones are much better at picking up CID though. And it was even faster before I added a wait(3) to try and improve CID success. When I first installed * it took till the third ring, watching the logs CLI scroll by showed me a mistake that changed it from 3 rings to 1 ring. I wish I remembered what it was. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 02/21/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Follow Me
Don't thank me, thank B J Weschke On 2/22/06, Max Clark [EMAIL PROTECTED] wrote: Thank You. On 2/21/06, C F [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. On 2/21/06, Max Clark [EMAIL PROTECTED] wrote: Hi all, I am interested in a follow me script for Asterisk - specifically I am looking for one that will prompt the calling party to record their name and then call through a list of numbers playing the recording. If a digit is pressed by the recipient then the call is put through. Is there anything like this available as an example for Asterisk? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools. At any rate... the config is basically one large XML file. I wrote my own interface to generate the XML files. We did get ahold of the provisioning tools to generate the XML files. Unfortunately, I don't think I'm allowed to give the tools out (some odd license). However, if you contact a distributor such as voipsupply, and tell them you plan to purchase XYZ amount, they may have some sort of deal they can work with Sipura/Linksys to get you the tools. I'd check there. If nothing else works, I don't think it would be illegal for me to send you an XML file for the SPA-2002. At that point you'd know what it is looking for and you could generate your own XML files. The only thing you couldn't do would be generate the 'sipura binary config files'... but you don't have to use those (As the XML works just fine). On 2/22/06, Gonzalo Servat [EMAIL PROTECTED] wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Try the Sipura SPA-2002.. at good prices from VoipSupply.com We have been using those now with 0 problems. We remote provision them from our office here. Once a minute (time configurable) each device checks in with us to check out its configuration file and see if it needs updates. The devices run around $60 a piece, so they are pretty cheap as well. RE the remote provisioning, did you have to pay some sort of license fee to get access to the tools to generate the remote provisioning configurations and instructions on setting it all up? I have 12 x PAP2-NA/SPA-2002 and changing one setting means going around and changing all the settings for each line of the 12 ATAs, that's 24 configuration changes in total - a real PITA. If you know of a way to obtain the tools to do the remote provisioning, I'd be grateful! Thanks, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints between servers?
Greetings all, Has anyone managed to get dialplan status hints working across multiple servers? I've separated a load of SIP users out across 2 servers today, but it'd be useful if they could still see each others' status. I've replaced the various hint lines for the sip devices now on another box with: exten = 210,hint,IAX2/otherserver/210 Where 210 is defined on the other server as follows: exten = 210,hint,SIP/210 All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
Correct. The XML works fine. If you need an example for the 2002, I will see if I can strip the information directly related to our company off and send it to you. Darrell S. Long BestWeb Corporation Matt wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools. At any rate... the config is basically one large XML file. I wrote my own interface to generate the XML files. We did get ahold of the provisioning tools to generate the XML files. Unfortunately, I don't think I'm allowed to give the tools out (some odd license). However, if you contact a distributor such as voipsupply, and tell them you plan to purchase XYZ amount, they may have some sort of deal they can work with Sipura/Linksys to get you the tools. I'd check there. If nothing else works, I don't think it would be illegal for me to send you an XML file for the SPA-2002. At that point you'd know what it is looking for and you could generate your own XML files. The only thing you couldn't do would be generate the 'sipura binary config files'... but you don't have to use those (As the XML works just fine). On 2/22/06, Gonzalo Servat [EMAIL PROTECTED] wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Try the Sipura SPA-2002.. at good prices from VoipSupply.com We have been using those now with 0 problems. We remote provision them from our office here. Once a minute (time configurable) each device checks in with us to check out its configuration file and see if it needs updates. The devices run around $60 a piece, so they are pretty cheap as well. RE the remote provisioning, did you have to pay some sort of license fee to get access to the tools to generate the remote provisioning configurations and instructions on setting it all up? I have 12 x PAP2-NA/SPA-2002 and changing one setting means going around and changing all the settings for each line of the 12 ATAs, that's 24 configuration changes in total - a real PITA. If you know of a way to obtain the tools to do the remote provisioning, I'd be grateful! Thanks, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and T38 Fax
See: http://bugs.digium.com/view.php?id=5090 Jorge Andy Kuo wrote: Hi, I tried to connect two T.38 capable SIP ATA's through Asterisk. I had canreinvite=yes and the 2 ATA's did talk directly to each other, but fax still failed. From Ethereal captures, I think the problem was when the originating ATA is ready to set up a T.38 session, it sends a message to the destination ATA to get ready for receiving T.38 packets. Unfortunately, Asterisk did not pass that message to the destination, so even though the actual T.38 packets did go directly to the destination, the destination ATA doesn't know it needs to switch to T.38 mode. What I am wondering is, if I can somehow configure the ATA's to be locked to T.38 only, no voice calls allowed, will it work? (since it is ONLY for receiving T.38 packets, we don't care if Asterisk pass on the message to the destination to switch to T.38) Any thoughts on whether this will work or this is just a crazy idea? Thanks. Andy On 2/21/06, Lee Howard [EMAIL PROTECTED] wrote: Carey Mould wrote: How can I get asterisk to work with faxes in my configuration? I have a WAN with Asterisk at the centre and Mediatrix 1104 gateways at the end nodes providing tone to legacy PBX's and fax machines. The Asterisk is connected to the PSTN via a Digium single port t1. The end nodes are connected via frame-relay 128kbps links. I want to use g.729 between the end nodes and the Asterisk box at the centre. TheMediatrix box supports T38. In my reading I am not seeing where asterisk supports T38. I believe that currently Asterisk SVN supports T.38 pass-through to some extent. Others here would be able to comment on that more fully. However, what you're seeking is T.38 gateway support, and Asterisk does not support that at the present. I know that there is some intention with spandsp to get T.38 gateway support there (so Asterisk would eventually support T.38 gateway via Unicall), but I suspect that's a long way off for your present needs. Again, I'm not really the one to be able to comment on this in detail. I'm not sure if the OpenH323 project (which I know supports at least T.38 pass-through) supports T.38 gateway, but even if it did, getting OpenH323 and Asterisk to work together may be difficult at best. I'm not really the one to be able to comment on that in detail, either. So for the time being I would consider that Asterisk does not support the T.38 features that you need to make this work and that you look at a different approach to getting your faxing working. 1) If a fax machine connected to the Meditrix box sends a fax to a location on the PSTN (legacy fax machine) where is the T38 converted back? That's what a T.38 gateway does. Asterisk doesn't do it. 2)Does Asterisk handle this? No. Not yet anyway. 3) How do I configure Asterisk to do this You can't presently. 4) What is the significance on the Digium web site where they have a CYA on the T1 board not supporting faxes ? You probably are referring to these support statements on virtually all of their products: The current state of faxing is incomplete and will not be supported. Well, it pretty much means what it says. I don't think you'll get any significant degree of support from Digium if you run into trouble with faxing using their hardware. All of that said, can you push a fax call through an Asterisk PBX? Yes. Search the archives and you'll see it can be done. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hints between servers?
Chris Bagnall wrote: All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? The latter... cross-server device state is not implemented. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
I had to drop 1.0.1.12 because it has a serious handset volume issue that seems to cut the handset volume in half. Fix one bug, cause another. Clint On 2/22/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Wed, 22 Feb 2006, Clint Sharp wrote: 2) GXP-2000: Not much better than the Budgetones, but at least the firmware [...[ that phone's quality).The speakerphone is useless due to echo issues. speakerphone echo bug was fixed in 1.0.1.12-Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 dialing trouble
Hi- Ihave a 7960 using chan_sccp which gives me a busy signal as soon as I dial the 1 in my string of 91555222. Can't figure out why. I do have a dialplan.xml file: DIALTEMPLATE TEMPLATE MATCH="*" Timeout="5"/ TEMPLATE MATCH="#..." Timeout="5"//DIALTEMPLATE Anyone have any insights? Thanks Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
On 2/22/06, Matt [EMAIL PROTECTED] wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools. This sounds like yet another reason to avoid purchasing Sipura equipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is SIP canreinvite working ok?
I've the following situation: Phone A: Codec GSM supported Phone B: Codec iLBC supported in sip.conf: [general] ... disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw canreinvite=yes ... (There's a lot of other SIP users, that's why I made the default codec list bigger than just GSM and/or ALAW) If phone A calls to phone B the conversation is established at SIP level, but there's no RTP traffic between the machines. If I make a sip show channels at the Asterisk console, I see: server*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.1.101phone_A 10095d01445 00103/0 ulaw No Tx: ACK 192.168.1.107phone_B 182E175F-F6 00102/2 ulaw No Tx: ACK 2 active SIP channels (ULAW?!?!?, not even ALAW!!!) As far as I understand, since in this case the communication can not be established directly between A and B (i.e. bypassing Asterisk as the media transport), given the fact that the A codec and the B codec are different, the REINVITE shouldn't be issued and discarded automatically for Asterisk, even and despite the fact canreinvite=yes is set. However, it seems to be issued anyway, so I can't hear anything. Am I doing something wrong, or this is effectively an Asterisk problem? Asterisk 1.2.4 SIP Client: SJPhone 1.60.289a I checked the REINVITE sent from Asterisk to the phones with Ethereal. Also, if I set canreinvite=no, the communication works nice, with GSM for one side and iLBC in the other. Thanks a lot for your attention. -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SV: Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail
Arne Morten Johansen wrote:
Thank you very much. For some reason emailsubject was not included in my example config. Well, it's working great now.
Last question, I promise :P. Is it possible to change the date format? I want
it in Norwegian.
; Set the date format on outgoing mails. Valid arguments can be found on the
; strftime(3) man page
;
; Default
emaildateformat=%A, %B %d, %Y at %r
; 24h date format
;emaildateformat=%A, %d %B %Y at %H:%M:%S
Hope this helps.
-Barry Flanagan
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 22. februar 2006 13:52
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: SV: Re: SV: Re: Fromstring when sending
e-mailonrecievedvoicemail
Arne Morten Johansen wrote:
It's fixed now.
Great!
In /etc/mail/ssmtp.conf, this (FromLineOverride=YES) line was commented
out. Removing that comment did the trick :)
Now I only need to change the e-mail's title. Is that possible?
Same way. In voicemail.conf: emailsubject and emailbody
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default
shows
; just the CIDNAME, if it is not null, otherise just the CIDNUM, or an
unknown
; caller, if they are both null.
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant
;
-Barry Flanagan
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Barry Flanagan
Sendt: 22. februar 2006 13:25
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Re: SV: Re: Fromstring when sending e-mail
onrecievedvoicemail
Arne Morten Johansen wrote:
As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not trusting the asterisk user. And the default behaviour of that is to not allow modification of the fromstring and serveremail. So if you have any idea how to fix that in Gentoo I would really appreciate it.
What package provides sendmail in Gentoo? Sendmail, Postfix, Exim are
all possibilities...
-Barry
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-Barry Flanagan
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Re: [Asterisk-Users] Best ATA for general residential deployment??
On 2/22/06, Darrell Long [EMAIL PROTECTED] wrote: Correct. The XML works fine. If you need an example for the 2002, I will see if I can strip the information directly related to our company off and send it to you. Hi Darrell, I would really appreciate it if you could send me the XML file (offlist), of course remove any company sensitive info. I have the SPA841 sample XML file and while it's a good base to start from, it has many SPA841 specific settings, so it would be better to get a copy of your 2002 one. This same XML file should work with a Linksys PAP2-NA, right? (it's basically the same device as a SPA-2002). If you don't know whether it works with one, I'll soon let you all know :) Thanks, Gonzalo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools. This sounds like yet another reason to avoid purchasing Sipura equipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a pick your poison situation in my opinion. Ht-486 anyone? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN interface cards with pass-through
Hi! Are there any multiport ISDN interface cards (PRI and BRI) which support pass-through in power-off mode. (I want to use Asterisk between the telco line and the existing PBX and I want pass-through when the power of the Asterisk server is switched off). regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problema calling from elesign h.323 to iax device
Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb 22 14:27:11 VERBOSE[22106] logger.c: -- IAX2/911-12 is ringing Feb 22 14:27:17 VERBOSE[22106] logger.c: -- IAX2/911-12 answered OH323/[EMAIL PROTECTED] Feb 22 14:27:18 VERBOSE[22105] logger.c: H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9', exception CTRL_ERROR (Capability Exchange [Rejected]). Feb 22 14:27:18 VERBOSE[22073] logger.c: -- H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9' cleared, reason 24 (Call ended with Q.931 cause [31 - Normal, unspecified]) Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'IAX2/911-12' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'dial' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'exten-vm' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'OH323/[EMAIL PROTECTED]' Any ideas? I'm using the channel_oh323.so module. I've another h.323 device tha works without problems. Best regards, -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problema calling from elesign h.323 to iax
Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb 22 14:27:11 VERBOSE[22106] logger.c: -- IAX2/911-12 is ringing Feb 22 14:27:17 VERBOSE[22106] logger.c: -- IAX2/911-12 answered OH323/[EMAIL PROTECTED] Feb 22 14:27:18 VERBOSE[22105] logger.c: H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9', exception CTRL_ERROR (Capability Exchange [Rejected]). Feb 22 14:27:18 VERBOSE[22073] logger.c: -- H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9' cleared, reason 24 (Call ended with Q.931 cause [31 - Normal, unspecified]) Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'IAX2/911-12' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'dial' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'exten-vm' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'OH323/[EMAIL PROTECTED]' Any ideas? I'm using the channel_oh323.so module. I've another h.323 device tha works without problems. Best regards, have you tried playing around with fastStart ; h245Tunnelling ; h245inSetup it helps to change and see what happens -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound problem sip chanel
I setup my aah box with a sip trunk at irisxa.iristel.net Incaming it is ok but when I try to dial 8 and the nr where I want to call I get all line is busy. In my log I have these: Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:19 VERBOSE[2721] logger.c: -- Accepting AUTHENTICATED call from 192.168.50.145: requested format = g729, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing Macro(IAX2/206-3, dialout-trunk|2|5149635279|) in new stack Feb 22 14:33:19 DEBUG[3239] pbx.c: _expression_ result is '1' Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing GotoIf(IAX2/206-3, 1?3:2)) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Goto (macro-dialout-trunk,s,3) Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing Macro(IAX2/206-3, user-callerid) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing DBget(IAX2/206-3, AMPUSER=DEVICE/206/user) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=206/user Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: set variable AMPUSER to 206 Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing DBget(IAX2/206-3, AMPUSERCIDNAME=AMPUSER/206/cidname) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=206/cidname Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: set variable AMPUSERCIDNAME to Cristian Paun Feb 22 14:33:19 DEBUG[3239] pbx.c: _expression_ result is '0' Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing GotoIf(IAX2/206-3, 0?5) in new stack Feb 22 14:33:19 DEBUG[3239] pbx.c: Not taking any branch Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetCallerID(IAX2/206-3, Cristian Paun 206) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing NoOp(IAX2/206-3, Using CallerID Cristian Paun 206) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing Macro(IAX2/206-3, record-enable|206|OUT) in new stack Feb 22 14:33:19 DEBUG[3239] pbx.c: Function result is '0' Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing GotoIf(IAX2/206-3, 0 0?2:4) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Goto (macro-record-enable,s,4) Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing AGI(IAX2/206-3, recordingcheck|20060222-143319|1140636799.15) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Feb 22 14:33:19 VERBOSE[3239] logger.c: recordingcheck|20060222-143319|1140636799.15: Outbound recording not enabled Feb 22 14:33:19 VERBOSE[3239] logger.c: -- AGI Script recordingcheck completed, returning 0 Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing NoOp(IAX2/206-3, No recording needed) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing Macro(IAX2/206-3, outbound-callerid|2) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing DBget(IAX2/206-3, USEROUTCID=AMPUSER/206/outboundcid) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=206/outboundcid Feb 22 14:33:19 VERBOSE[3239] logger.c: -- DBget: set variable USEROUTCID to Feb 22 14:33:19 DEBUG[3239] pbx.c: _expression_ result is '0' Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing GotoIf(IAX2/206-3, 0?4) in new stack Feb 22 14:33:19 DEBUG[3239] pbx.c: Not taking any branch Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetCallerID(IAX2/206-3, K2 Systems Inc.) in new stack Feb 22 14:33:19 DEBUG[3239] pbx.c: _expression_ result is '1' Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing GotoIf(IAX2/206-3, 1?6) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Goto (macro-outbound-callerid,s,6) Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing NoOp(IAX2/206-3, CallerID set to K2 Systems Inc.) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetGroup(IAX2/206-3, OUT_2) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing CheckGroup(IAX2/206-3, 2) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetVar(IAX2/206-3, DIAL_NUMBER=5149635279) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetVar(IAX2/206-3, DIAL_TRUNK=2) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing AGI(IAX2/206-3, fixlocalprefix) in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Feb 22 14:33:20 VERBOSE[3239] logger.c: -- AGI Script fixlocalprefix completed, returning 0 Feb 22 14:33:20 VERBOSE[3239] logger.c: -- Executing SetVar(IAX2/206-3, OUTNUM=85149635279) in new stack Feb 22 14:33:20 VERBOSE[3239] logger.c: -- Executing Cut(IAX2/206-3, custom=OUT_2|:|1) in new stack Feb 22 14:33:20 WARNING[3239] ast_expr2.y: non-numeric argument Feb 22 14:33:20 DEBUG[3239] pbx.c: _expression_ result
Re: [Asterisk-Users] Best ATA for general residential deployment??
On 2/22/06, Martin Joseph [EMAIL PROTECTED] wrote: On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: This sounds like yet another reason to avoid purchasing Sipura equipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a pick your poison situation in my opinion. Ht-486 anyone? Yeah, unfortunately it is. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
On Wed, 22 Feb 2006, Matt wrote: Unfortunately, I don't think I'm allowed to give the tools out (some odd license). However, if you contact a distributor such as voipsupply, and tell them you plan to purchase XYZ amount, they may have some sort of deal they can work with Sipura/Linksys to get you the tools. I'd check there. The information needed for XML provisioning is openly available from sipura/linksys. The actual linksys provisioning tools may be under some license but the XML provisioning syntax is not. It is actually ridiculously simple. I'm writing a php script which can provision equipment out of the box, totally plug and play. Right now it can configure spa3000, snom 360, and gxp2000. I'll be adding polycoms to the list soon. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best ATA for general residential deployment??
On 2/22/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: [snip] The information needed for XML provisioning is openly available from sipura/linksys. The actual linksys provisioning tools may be under some license but the XML provisioning syntax is not. It is actually ridiculously simple. [snip] The only XML info I found on the Sipura site was for the SPA 841. Nothing on the ATAs. Have looked on the Linksys site too and no info at all on it. Do you have a link? Cheers, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What are these error messages in my logs?
Hello, I am getting a bizzare amount of error messages in the log files. The system seems to be running fine...no one is reporting any issues and all calls are coming and going. System is showing higher than average memory usage. eth0 is showing a high number of errors Running v1.1 Has happened on older versions and have been seeing this for quite some time but have just now asked if anyone knows what is going on. Could not find anything in wiki about it. Thanks for any comments. debug below: Feb 22 14:33:41 DEBUG[18472] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 22 14:33:44 DEBUG[18472] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 22 14:33:44 DEBUG[18472] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 22 14:33:46 DEBUG[18472] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 22 14:33:54 DEBUG[18472] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 22 14:33:54 DEBUG[18472] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 22 14:33:55 DEBUG[18472] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found and it just continues on and on ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
