Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
Michael Loftis wrote: The fact that it crashes says something more is wrong naywa...have you tried running anything like Memtest86 on the box fora time just to see if it comes up with anything? Yes I have allowed memtest86 to run for about 9 hours overnight... that is one of the first things I tried in fact. Little tireds now so you may have already done all this but make sure you have latest libpri, zaptel, and asterisk, in that order. Yup, every day or two I update zaptel and asterisk (don't have any PRIs yet), just hoping that will solve the problem. So far not working out too well :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Netiquette, newbies, politeness and such (was G.729 . . . I SMELL SMOKE!)
[EMAIL PROTECTED] wrote: On Sat, 2004-10-23 at 01:06 -0400, Jim Van Meggelen wrote: Few will disagree that the careful application of netiquette will be a benefit to any newsgroup/mailing list/board; and top posting is something that should be used sparingly. Nevertheless, top posting is not the horrid crime some might have us believe. When used appropriately, it serves very well, and only causes offense to the ideologues. Me too-type top posing is usually of no benefit, but when someone is commenting on a tangled and involved thread, it can make sense to frame the entirety of the thread in a thoughtful top post. Don't forget the same people who refuse to trim the bottom of the post and we end up with 20(your case only 1) copy of the mailing list footer. Sure, but then do we want to start picking on grammar and spelling as well? That's something that drives me nuts, yet I realize that many people consider it to be unimportant. It was a hotly debated topic in Usenet for some years, until it was realized that the community was not served by all of this endless bickering about grammar and punctunation. Many people fear, however, that eventually we wi11 |\|0+ b3 4b13 +0 u|\|d3r5+4|\|d 34(|-| 0+|-|3r 4|\|ym0r3 (thanks to http://www.computerhope.com/jargon/l/leetspea.htm for the translation to leetspeak/133+5p34k). I also consider long, fancy signatures to be needless, but I respect people's right to have them. I see no value in making an issue out of it. Then we get to the most dangerous beast, the abusive, expert troll. This is someone who clearly is very intelligent and articulate, and could argue their value due to a) their willingness to contribute, b) their level of knowledge and c) their fantastic writing skills. Unfortunately, these folks reduce their value to almost nothing by virtue of their pathetic lack of any manners whatsoever. They will drive away more people than they help -- and that doesn't bother them in the slightest. What a waste of talent. As I am sure to be painted by the above brush, let me offer just a small point here. I have had just a bit of time to think this over after politely listenening to the same argument from another person this weekend. You seem to not realize that those who are knowlegable are only so due to the vast amount of time we put into learning. Not at all. I respect that truth. On the other hand, I also know that some people are able to attain knowledge easier than others -- their minds simply absorb knowledge more efficiently. People who are less capable in this regard know this, and may prefer to obtain their answers from someone they consider an expert -- rather than do reserch on their own. Is this an offense; or is it a compliment? I'm sure there are many people who are like me and are trying to spend a lot of time learning several projects that have no overlap. While we seek all this knowlege, I hope the others like me actually try and do things outside of the computer world as well. LOL! Well said! Now I want you to realize that many of the really newbie or lazy (these are NOT equal in the level I detest) questions that are answerable with a quick browse of the wiki or a simple google search end up being equivalent to SPAM in my mailbox as I try and search for information that furthers my knowlege. Understand that I learn from looking at what others are doing, and answers to others questions. Here is a quote that appeared on the mailing list today. It is a profound testament to the newbie's angst: ... it's not that I have not been reading (ask my wife how many nights I have slept in the last week), and it's not that there is not a huge amount of info out there. The problem I am having is finding the info I need in any sort of organized way. Many newbies share this poster's misery, though few may be able to articulate it as well. Should they be flamed for that? I say no. When it comes to problems of *any* sort, people seem to approach them with one of these two (very different) mindsets: The first type try to find someone, anyone, to take their problems away (or at least to blame). The second, however, take ownership of the problem(s), and accept the responsibility for solving same. When people from the first group post messages, you are quite correct if you assume that your efforts on their behalf will be of little benefit - these folks are generally looking to take, not give. But they don't really matter; it's the people from the second group who we need to be aware of, because when you answer their questions, not only will your answers be used to fullest effect, but it is also quite likely that they will in turn pass that knowledge on in kind. You will have sown a seed. Who can say what it will grow into? Once you identify someone from the first group, you may want to gently probe and determine if this condition is cureable or not. If not, then save bandwidth and ignore them. Do not flame them, nag
RE: [Asterisk-Users] KSS/BLF on Asterisk
[EMAIL PROTECTED] wrote: On Mon, 2004-10-25 at 14:05, Paul Hales wrote: First you need to set up the hint function in extensions.conf: exten = 6003,hint,SIP/6003 Then set the SNOM's buttons (FUNCTION KEYS) to be speed dials (DESTINATION) to the extensions in question. Has anyone managed to do this with a polycom IP phone? eg, the IP500/IP600 phones? And I'm starting to wonder if 3com's phones need another look. They claim SIP compliance, and their 3100-series sets look very interesting (aspecially the 3105 attendant console). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] KSS/BLF on Asterisk
Thanks very much. I'll have to add some snom phones to my lab. [EMAIL PROTECTED] wrote: First you need to set up the hint function in extensions.conf: exten = 6003,hint,SIP/6003 Then set the SNOM's buttons (FUNCTION KEYS) to be speed dials (DESTINATION) to the extensions in question. That's it! Regards, PaulH -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Monday, 25 October 2004 12:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk I have it working under Asterisk, and it's very good. I will post the how-to when I get back from Lunch! Regards, PaulH -Original Message- From: Henry Devito [mailto:[EMAIL PROTECTED] Sent: Monday, 25 October 2004 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk I am buying a Snom phone this week. I will play with this feature and see what I can get going. I will share my findings. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Sunday, October 24, 2004 6:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk [EMAIL PROTECTED] wrote: Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. I've looked all over the wiki, and all the documentation I could get my hands on, Where did you find anything about the hint priority? I am interested in trying to make this work. The only references I could find to it were here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20standa rd%20exten sions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom and here: http://www.mail-archive.com/[EMAIL PROTECTED]/ms g49781.htm l Not much to work from, but the lack of documentation on this feature is probably signifigant. I realize that this is going to be something that requires more research on my part, as no one appears to be using it very much. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Several FXS Ports
Duane Cox [EMAIL PROTECTED] lazily top-posted: Is there not a search engine for the mailing lists? And before some jerk responds with google, google isn't the best solution. What's wrong with Google? Google will not only find you the mail list articles you need, but might even scare up a few other links that could be of interest - such as to the WIKI. The only problem with Google is that the frequency of updates means that recent articles might not be indexed yet. There's no reason why you can't search your own local mail list archive for recent articles and then follow that up with a Google search for older articles. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACT Gateways
On Mon, 25 Oct 2004 16:55:55 +1300, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I would say Asterisk would work with any SIP device out there, it would depend upon your individual configurations. As long as the device SIP/IAX/H323 it should be straight forward to get it plugged into Asterisk. Gettting plugged in perhaps, but work is a different matter. I have seen it many times that certain SIP devices needed a firmware upgrade to work with certain other devices, including Asterisk. For example, the Hitachi Cable WIP-5000 WiFi SIP phones only worked with Asterisk at a meeting with Hitachi Cable here in Tokyo *after* their technicians upgraded the firmware. Then when we got two test units, apparently with different firmware, they didn't work. Some time ago ACT's SIP phones showed some really weird behaviour with older Asterisk versions, ie you could only dial once and then the phone would lock up and needed a reboot. Although this was only with older versions of Asterisk and everything worked fine with more recent versions, ACT fixed this, just to make sure. Sometimes it is very small things. Say, music on hold features or message waiting indicators etc etc etc. If such features don't work properly, you can still use the phones, but you would really want to use the full feature set. So, it is indeed quite helpful if a vendor takes Asterisk serious enough to look into any interoperability issues that might pop up. As far as ACT are concerned, I am very happy with their commitment and attitude towards Asterisk. Of course many other vendors will by now also have recognised Asterisk as a significant platform that they should support without any buts and ifs, but this is still not something you can take for granted, as the Hitachi example shows. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-OH323 Invalid format RTP
On Saturday 23 October 2004 05:08, M. Ehsanul Karim wrote: I am trying to connect asterisk to a H323 gateway , the call rings and connects perfectly , but there seems to be no audio. It gives out this error: Invalid format of RTP addresses. Please let me know, what may go wrong here. You have incorrect codec settings. What is set on client doesn't match oh323 codec settings. Dmitry Thanks. Ehsanul Karim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iLBC/PCM16 Huge Cost
Senad Jordanovic wrote: Trevor Peirce wrote: SNIP Trevor, You are better off using this instead of mpg321: http://bugs.digium.com/bug_view_page.php?bug_id=0002379 Mark... Any chance including this patch in the 1.0? NO! 1.0 is a *STABLE* branch. *ONLY* bug fixes should go into Stable branches. If you meant the next release (likely called 1.1), then yes I agree that it should be included. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I have Asterisk Hylafax on a server. What else do I need...?
Hi everyone! I have an Asterisk server here that also has Hylafax installed on it. What else do I need to have that server send/receive faxes? Regards, Evert Meulie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACT Gateways
On Sun, 24 Oct 2004 18:11:44 -0600, Joseph [EMAIL PROTECTED] wrote: Has anybody tested any gateways from ACT Last time those gateways came up in a conversation it was concluded that efforts should be concentrated on the phones so as to not dilute engineering resources with too many things. This was about 2 months ago or so. I have come to read this as: gateways are not ACT's priority -- phones are. I would therefore recommend to look for gateways elsewhere, for example Mediatrix. rdgs benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy setup
I haven´t tougth about being re-provisioning the iaxy box :)... But how do you detect the dns change? wich ddns company are u using? Take look at dyndns.org, they have both free and paid services. I set a cron job to run every 10 minutes or so and based on a script I found at DDNS, it detects its ip address and if changed does a bunch of things like notify DDNS so the DNS is updated (some Linksys and other NAT routers will do this btw) and change the ip in the sip.conf, do a sip reload, reprovision the IAXy. works like a charm! Here is their list of clients, many of which are just scripts in Perl, PHP etc. https://www.dyndns.org/services/dyndns/clients.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How useful is the screen on IP phones?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Paul Crick wrote: | Hey Jay | | All the stuff you've described is possible. I've done some playing | with the XML services on a Cisco 7960 to give ACD queue stats and | system uptime info. The phone has a mini web browser built in so | it's pretty easy to knock up some glue scripts in the back end to | do what you want to do. I think there are some examples on the wiki | too. | | Cheers Paul | Hi I'm using 14 cisco 7940 as Dynamic queue agents. They use the pixel based screen, to login/logout from queues. They can also see the queues stats. | ___ Asterisk-Users | mailing list [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Manuel João S. Costa Amaro [EMAIL PROTECTED] ICQ: 57398499 MSN: [EMAIL PROTECTED] As únicas pessoas que aprecio são os loucos: os que são loucos para viver, loucos para falar, loucos para se salvar, desejosos de tudo ao mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que ardem, se inflamam e brilham como fabulosos fogos-de-artifício. (Jack Kerouac) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBfL+DJUm/Bor63CERAt5RAJ47cfoPagbwekr5mHGoc9Vea0kB1wCeNr9B wSf6jO6t78PAEBkZLue2Im0= =2wmf -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Netiquette, newbies, politeness and such (was G.729 . . . I SMELL SMOKE!)
snip referring to long nights of reading Many newbies share this poster's misery, though few may be able to articulate it as well. Should they be flamed for that? I say no. Let me add (as a newbie with many projects and a life and a production asterisk pbx) that seeing the constant response of look it up on the mailing list on the IRC channel and this has been discussed many time here so don't be lazy in the list itself is one thing. But when you do a lot of searching, you'll see that a large amount of messages brought up in a search will be empty of any useful content, not because the person posting couldn't answer the question, but because they refused to do so and then complained about the noise. IMO, if you want to bitch, answer the question, *then* bitch! If you don't have an answer or don't want to give one, use your right to remain silent instead of adding more useless search results. Complaining about the noise in a question thread (of which this current one is not) is just ADDING MORE noise. The community has a clear choice, either attract people who are just getting started (which I'd like to see, we *need* people other that programmers, this is the *users* list) or repel them and keep asterisk in the sorcery domain that unix stayed in for centuries (well, the equivalent of in technical temporal frame.) I don't doubt that there are people who like that idea, but it won't advance the cause of asterisk at all, merely keep it in the ghetto. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto get voicemail $VM_ vars into externnotify script?
I am trying to slap together a script that will email2sms the details of
the voicemails left on my * box to my gsm phone. I can't figure out how
to get my script to pick up the voicemail vars like ${VM_MSGNUM},
${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}.
change the voicemail.conf to include them in the body using the emailbody as in
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have received a ${VM_DUR}
message from ${VM_CALLERID}. The message was left on ${VM_DATE}.
The above probably needs to be on one line. Just change the wording
the way you want it.
___
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[Asterisk-Users] CDR Dokumentation
ist there any cdr dokumentaion about the cdr format? thx! thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
amount of info out there. The problem I am having is finding the info I need in any sort of organized way. Here is an excellent intro to asterisk that treats ALL the important issues when you first start using asterisk: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Watch for typos in these articles and beware of the fact that they may not be up to date, but that one certainly covers what you are looking for. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip users registering fails
Hi. I'm using * version 1.0.1 I have 3 sip users in my sip.conf file. Everything was worked fine until today. One extension remains registered when the client phone isn't running and now I can't register with that terminal. I've tried to restart client computer and the server itself but the problem remains. My clients are X-PRO. This message appears in server debug: Oct 25 11:09:01 WARNING[1093143472]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) this is where I see the problem:ser9902p468*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1516/1516(Unspecified)D 255.255.255.255 0Unmonitored 1530/153010.245.201.152 D 255.255.255.255 5060 Unmonitored 1426/1426(Unspecified)D 255.255.255.255 0Unmonitored 1530 extension isn't running Anyone could help me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum A800 OH323 problem
Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ACT Gateways
Dear Benjamin, Unfortunately the Mediatrix products are very expensive. For example the price for Two-port access device with SIP protocol is $275. The price for the same of good looking Gateways from Yoda (www.yoda.com.tw) is $85. Unfortunately I haven't experience with Yoda devices. Also they required min. 10 pcs for ordering. Best Regards, Miroslav Nachev BoAML On Sun, 24 Oct 2004 18:11:44 -0600, Joseph BoAML [EMAIL PROTECTED] wrote: Has anybody tested any gateways from ACT BoAML Last time those gateways came up in a conversation it was concluded BoAML that efforts should be concentrated on the phones so as to not dilute BoAML engineering resources with too many things. This was about 2 months BoAML ago or so. I have come to read this as: gateways are not ACT's BoAML priority -- phones are. I would therefore recommend to look for BoAML gateways elsewhere, for example Mediatrix. BoAML rdgs BoAML benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Dokumentation
see: http://www.voip-info.org/wiki-Asterisk+billing - shabanip - Original Message - From: Thomas Kuepper [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 12:42 PM Subject: [Asterisk-Users] CDR Dokumentation ist there any cdr dokumentaion about the cdr format? thx! thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Dokumentation
also: http://www.voip-info.org/wiki-Asterisk+cdr+csv - Original Message - From: Thomas Kuepper [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 12:42 PM Subject: [Asterisk-Users] CDR Dokumentation ist there any cdr dokumentaion about the cdr format? thx! thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] protection
hi, how do u prevent unauthorized usage or block users temporarily to use Asterisk services ? Is defaultip and secret enought ? what u do to prevent this. tia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip users registering fails
I have this same problem but only noticed it when I moved one of my phones from my house to the office. When the Internet connection changed the firewall setup changed between the phone and our Asterisk machine so I just assumed it was a firewall issue at the office. I get this error for this one extension but the phone still operates perfectly, does your phone work or no? -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 4:23 AM Subject: [Asterisk-Users] sip users registering fails Hi. I'm using * version 1.0.1 I have 3 sip users in my sip.conf file. Everything was worked fine until today. One extension remains registered when the client phone isn't running and now I can't register with that terminal. I've tried to restart client computer and the server itself but the problem remains. My clients are X-PRO. This message appears in server debug: Oct 25 11:09:01 WARNING[1093143472]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) this is where I see the problem:ser9902p468*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1516/1516(Unspecified)D 255.255.255.255 0 Unmonitored 1530/153010.245.201.152 D 255.255.255.255 5060 Unmonitored 1426/1426(Unspecified)D 255.255.255.255 0 Unmonitored 1530 extension isn't running Anyone could help me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] protection
If they are using SIP devices you can just change the secret to some value they will not know, this should work at least for outgoing calls, not sure for incoming. Just tried this earlier and once I changed the secret I could no longer make calls. -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 4:47 AM Subject: [Asterisk-Users] protection hi, how do u prevent unauthorized usage or block users temporarily to use Asterisk services ? Is defaultip and secret enought ? what u do to prevent this. tia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware
On Sunday 24 October 2004 05:15, Steve Underwood wrote: Stay away from boards with Intel chipsets. Those are problematic in my experience. The FX, LX, 820, 840 and various others have been extremely flaky, and caused no end of problems. :-) VIA used to be bad, but seem to get steadily better. Intel are just erratic. I think most makers have made good and bad chipsets. Go with known good chips, not specific makers. The same goes with motherboards. So what are known good chips ?? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3com with Asterisk
Hi Lisa Is ther eany more I need to know about using 3com with Asterisk? Other than allowing ulaw. Thanks Doug Reid Director Stormcorp Network Solutions (Pty) Ltd Tel:+27 11 807 1141 Fax:+27 11 807 3504 Mobile: +27 83 989 0008 E-Mail: [EMAIL PROTECTED] Web:www.stormcorp.co.za --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message, you are hereby notified that you must not disseminate, copy or take any action in reliance on it. If you have received this message in error, please notify Stormcorp Network Solutions, its subsidiaries or associates, immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Stormcorp, its subsidiaries or associates. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AST doesn't start after update from 0.5 to 1.0
Please be graceful if I might sound foolish, it's my first posting here. I was happy to have asterisk 0.5 running in a very limited one-phone test configuration. Then I downloaded 1.0 and installed it. Afterwards, asterisk didn't even start any more. The messages are: Notice: iax2-provision.c:496 iax_provision_reload: no iax provisioning configuration found, IAX provisioning disabled. Warning: phx.c:2304 ast_register'_application: Already hvae an application 'Voicemail2' Warning: phx.c:2304 ast_register'_application: Already hvae an application 'VoicemailMain2' Warning: loader.c:334 ast_load_resource: app_voicemail.so: load_module failed, returning -1 Warning: loader.c:429 load_modules: loadin module app_voicemail.so failed! Any help is highly appreciated ! Thank you, Joe. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwdith usage
Does anybody know if the voice actually gets routed through Asterisk for callsbetween SIP devices? I just wonder if calls between SIP devices would take up any bandwidth or CPU at the Asterisk server. Please advise. Thanks, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwdith usage
Joseph Shi [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Does anybody know if the voice actually gets routed through Asterisk for calls between SIP devices? I just wonder if calls between SIP devices would take up any bandwidth or CPU at the Asterisk server. Please advise. SIP devices will send re-invitations in an effort to find the most efficient route for the voice data, bypassing the server(s) etc. In a lot of cases, the two endpoints will end up speaking to one another directly. You can set up Asterisk to keep itself in the loop (canreinvite = no), or it might want to remain in the loop regardless of your settings. For instance, Asterisk will want to remain in the loop if you're recording the call - for obvious reasons. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ACT Gateways
On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev [EMAIL PROTECTED] wrote: Unfortunately the Mediatrix products are very expensive. just one example. my point was that as of this moment, ACT are more focussed on their phones and it may well be wise to look for gateways elsewhere for the time being, whereever that elsewhere may be. example the price for Two-port access device with SIP protocol is $275. I don't really understand the obsession with FXS devices. The only uses I see for FXS are - connect a FAX machine, where FAX may not be the best application for VoIP anyway, - connect an existing cordless phone, where you probably have only one such device and a Grandstream HT286 will just do fine, - connect the analog phone in a hotel to a travel adapter, IAXy would seem to be the best choice here because you are so much more likely to encounter NAT traversal problems and other obstacles that you may not be able to resolve with a SIP device, - feed some Internet based phone services into a legacy PBX that wants to see them as CO lines, here again, depending on the number of feeds, HT286 may be cheap and cheerful enough. For anything else IP phones should be the default with no buts and no ifs. I am always puzzled by how people desperately hang on to legacy stuff they don't really need and in the process create a beast of a kludge technology. The x86 architecture (or lack thereof) should be an example that serves to show how not to design your stuff with legacy support as your all-overriding number one priority. So, let's not make the same mistake with VoIP. Let's get rid of analog phones as fast and forcefully as we possibly can. In other words, FXS should be the very very last resort when there is really no other way. Having said that, I notice that Yoda have a 4 port FXO gateway (VG400), or at least it can be configured to be a 4 port FXO gateway. Now, that is rather interesting. Do you have any idea how much this device costs (ballpark figure wise) and how well it can adapt to PSTNs in other countries? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] ACT Gateways
Hi, We choose the Mediatrix 2102 with 2 analogue and 2 ethernet ports. Cost: £89.99 (roughly equiv $165). We are using these to hook up Faxes and DECT phones (cordless). The top of the range business DECT from from BT is £30 (if you buy a few from trade). Worth mentioning that even VoiceMail indication works on the BT analogue phone. Also the voice quality was actually better on the top of the range business DECT phone than the top of the range home BT phone which retails at around £90 (the one that includes SMS / mobile sim card support). What other cordless choices are there for native SIP phones??? Zyxel Prestige 2000W Wireless SIP Phone = £159.99 (on sale even). I think you can easily do the math and realise what the best option is. HTH Alex -Original Message- From: Benjamin on Asterisk Mailing Lists [mailto:[EMAIL PROTECTED] Sent: 25 October 2004 11:32 To: Miroslav Nachev Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Re[2]: [Asterisk-Users] ACT Gateways On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev [EMAIL PROTECTED] wrote: Unfortunately the Mediatrix products are very expensive. just one example. my point was that as of this moment, ACT are more focussed on their phones and it may well be wise to look for gateways elsewhere for the time being, whereever that elsewhere may be. example the price for Two-port access device with SIP protocol is $275. I don't really understand the obsession with FXS devices. The only uses I see for FXS are - connect a FAX machine, where FAX may not be the best application for VoIP anyway, - connect an existing cordless phone, where you probably have only one such device and a Grandstream HT286 will just do fine, - connect the analog phone in a hotel to a travel adapter, IAXy would seem to be the best choice here because you are so much more likely to encounter NAT traversal problems and other obstacles that you may not be able to resolve with a SIP device, - feed some Internet based phone services into a legacy PBX that wants to see them as CO lines, here again, depending on the number of feeds, HT286 may be cheap and cheerful enough. For anything else IP phones should be the default with no buts and no ifs. I am always puzzled by how people desperately hang on to legacy stuff they don't really need and in the process create a beast of a kludge technology. The x86 architecture (or lack thereof) should be an example that serves to show how not to design your stuff with legacy support as your all-overriding number one priority. So, let's not make the same mistake with VoIP. Let's get rid of analog phones as fast and forcefully as we possibly can. In other words, FXS should be the very very last resort when there is really no other way. Having said that, I notice that Yoda have a 4 port FXO gateway (VG400), or at least it can be configured to be a 4 port FXO gateway. Now, that is rather interesting. Do you have any idea how much this device costs (ballpark figure wise) and how well it can adapt to PSTNs in other countries? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Geotel integration with Asterisk
Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. Their greatest claim to fame is that their peripheral monitor PC sits on your premise, and connects to your brand x pbx to report upstream to the telco router (actually a redundant pair PC) as to the ingoings of your call centre. The decision to terminate the call on a particular call centre is done in the telco cloud at the SS7 layer. Each call centre has 250ms to respond to the correct status or the telco default-routes the call based on the tables in the NAM. This feature is self-healing dynamic routing. Proactive rather than reactive when your call volumes change or a failure takes a centre offline/snow storm means only half of your agents show up today in one area of the country, etc. It allows a translation between disparate PBX's to participate in this scheme so it is a huge boon in mergers/acquisitions. Just drop this Peripheral Monitor (pair) in your CC and you are intergrated into our enterprise. Actually reporting is one of the weakest links in the Geotel (now Cisco ICM (Intelligent Call Manager)) platform. Countless clients complain about this and at their user conference they even came out and admitted it. The data elements are there, but they don't have a good handle on how to rationalize them. Bell Canada, Allstream, MCI and ATT offer this now that I am aware of. Yes it is very expensive, but for multi-site high-availability services like banks, airlines and insurance companies it pays off in spades. dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy setup
[EMAIL PROTECTED] wrote: I haven´t tougth about being re-provisioning the iaxy box :)... But how do you detect the dns change? wich ddns company are u using? Take look at dyndns.org, they have both free and paid services. I set a cron job to run every 10 minutes or so and based on a script I found at DDNS, it detects its ip address and if changed does a bunch of things like notify DDNS so the DNS is updated (some Linksys and other NAT routers will do this btw) and change the ip in the sip.conf, do a sip reload, reprovision the IAXy. works like a charm! I'm not getting how the IAXy obtains the new information. Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ACT Gateways
Dear Benjamin, I think that the combination of FXS with FXO ports is very useful for the SOHO and Medium Enterprises. Also the prices of Yoda and other Asian companies are very suitable (~$40 per FX? port) and is better than GrandStream ATA. The other future that is extra than GrandStream is WAN/LAN ports and Router/NAT possibilities. Yes, I now all prices of Yoda, and I am looking for some partners with which to combine one order for samples. The problem is that they require min. 10 pcs per order. The price for samples of VG400 is $320. The device use TI DSP for coding. The prices for regual (big) quantities is $60 per port. Best Regards, Miroslav Nachev BoAML On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev BoAML [EMAIL PROTECTED] wrote: Unfortunately the Mediatrix products are very expensive. BoAML just one example. my point was that as of this moment, ACT are more BoAML focussed on their phones and it may well be wise to look for gateways BoAML elsewhere for the time being, whereever that elsewhere may be. example the price for Two-port access device with SIP protocol is $275. BoAML I don't really understand the obsession with FXS devices. BoAML The only uses I see for FXS are BoAML - connect a FAX machine, where FAX may not be the best application for BoAML VoIP anyway, BoAML - connect an existing cordless phone, where you probably have only one BoAML such device and a Grandstream HT286 will just do fine, BoAML - connect the analog phone in a hotel to a travel adapter, IAXy would BoAML seem to be the best choice here because you are so much more likely to BoAML encounter NAT traversal problems and other obstacles that you may not BoAML be able to resolve with a SIP device, BoAML - feed some Internet based phone services into a legacy PBX that wants BoAML to see them as CO lines, here again, depending on the number of feeds, BoAML HT286 may be cheap and cheerful enough. BoAML For anything else IP phones should be the default with no buts and no BoAML ifs. I am always puzzled by how people desperately hang on to legacy BoAML stuff they don't really need and in the process create a beast of a BoAML kludge technology. The x86 architecture (or lack thereof) should be an BoAML example that serves to show how not to design your stuff with legacy BoAML support as your all-overriding number one priority. So, let's not make BoAML the same mistake with VoIP. Let's get rid of analog phones as fast and BoAML forcefully as we possibly can. BoAML In other words, FXS should be the very very last resort when there is BoAML really no other way. BoAML Having said that, I notice that Yoda have a 4 port FXO gateway BoAML (VG400), or at least it can be configured to be a 4 port FXO gateway. BoAML Now, that is rather interesting. Do you have any idea how much this BoAML device costs (ballpark figure wise) and how well it can adapt to PSTNs BoAML in other countries? BoAML rgds BoAML benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ACT Gateways
Dear Alex, From where you found this device for $165? I found that the List Price of this device is $220. Can you send me the URL or some contacts? Best Regards, Miroslav Nachev AB Hi, AB We choose the Mediatrix 2102 with 2 analogue and 2 ethernet ports. AB Cost: £89.99 (roughly equiv $165). AB We are using these to hook up Faxes and DECT phones (cordless). AB The top of the range business DECT from from BT is £30 (if you buy a few from trade). AB Worth mentioning that even VoiceMail indication works on the AB BT analogue phone. Also the voice quality was actually better on AB the top of the range business DECT phone than the top of the range AB home BT phone which retails at around £90 (the one that includes AB SMS / mobile sim card support). AB What other cordless choices are there for native SIP phones??? AB Zyxel Prestige 2000W Wireless SIP Phone = £159.99 (on sale even). AB I think you can easily do the math and realise what the best option is. AB HTH AB Alex AB -Original Message- AB From: Benjamin on Asterisk Mailing Lists AB [mailto:[EMAIL PROTECTED] AB Sent: 25 October 2004 11:32 AB To: Miroslav Nachev AB Cc: Asterisk Users Mailing List - Non-Commercial Discussion AB Subject: Re: Re[2]: [Asterisk-Users] ACT Gateways AB On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev [EMAIL PROTECTED] wrote: Unfortunately the Mediatrix products are very expensive. AB just one example. my point was that as of this moment, ACT AB are more focussed on their phones and it may well be wise to look AB for gateways elsewhere for the time being, whereever that AB elsewhere may be. example the price for Two-port access device with SIP protocol is $275. AB I don't really understand the obsession with FXS devices. AB The only uses I see for FXS are AB - connect a FAX machine, where FAX may not be the best application for VoIP anyway, AB - connect an existing cordless phone, where you probably have AB only one such device and a Grandstream HT286 will just do fine, AB - connect the analog phone in a hotel to a travel adapter, AB IAXy would seem to be the best choice here because you are so much AB more likely to encounter NAT traversal problems and other AB obstacles that you may not be able to resolve with a SIP device, AB - feed some Internet based phone services into a legacy PBX AB that wants to see them as CO lines, here again, depending on the AB number of feeds, HT286 may be cheap and cheerful enough. AB For anything else IP phones should be the default with no AB buts and no ifs. I am always puzzled by how people desperately AB hang on to legacy stuff they don't really need and in the process AB create a beast of a kludge technology. The x86 architecture (or AB lack thereof) should be an example that serves to show how not to AB design your stuff with legacy support as your all-overriding AB number one priority. So, let's not make the same mistake with AB VoIP. Let's get rid of analog phones as fast and forcefully as we AB possibly can. AB In other words, FXS should be the very very last resort when there is really no other way. AB Having said that, I notice that Yoda have a 4 port FXO AB gateway (VG400), or at least it can be configured to be a 4 port AB FXO gateway. Now, that is rather interesting. Do you have any idea AB how much this device costs (ballpark figure wise) and how well it AB can adapt to PSTNs in other countries? AB rgds AB benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwdith usage
I also wanted to add to this: If you have users behind NATs then the canreinvite=yes will essentially make the phone ring but when it's picked up the call will break up and the two parties can't talk. Just went through this in my setup, the only way to get the two sides to talk was to set canreinvite=no. This may be because I had both the calling and called sip devices behind a NAT on each end of the call. It may work if only one end is behind a NAT??? -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 5:30 AM Subject: RE: [Asterisk-Users] Bandwdith usage Joseph Shi [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Does anybody know if the voice actually gets routed through Asterisk for calls between SIP devices? I just wonder if calls between SIP devices would take up any bandwidth or CPU at the Asterisk server. Please advise. SIP devices will send re-invitations in an effort to find the most efficient route for the voice data, bypassing the server(s) etc. In a lot of cases, the two endpoints will end up speaking to one another directly. You can set up Asterisk to keep itself in the loop (canreinvite = no), or it might want to remain in the loop regardless of your settings. For instance, Asterisk will want to remain in the loop if you're recording the call - for obvious reasons. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On October 24, 2004 10:27 am, Joe Greco wrote: would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. You're absolutely right, and I apologize. I thought she worked at Digium. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwdith usage
Thanks for the info. - Original Message - From: Me [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 6:52 PM Subject: Re: [Asterisk-Users] Bandwdith usage I also wanted to add to this: If you have users behind NATs then the canreinvite=yes will essentially make the phone ring but when it's picked up the call will break up and the two parties can't talk. Just went through this in my setup, the only way to get the two sides to talk was to set canreinvite=no. This may be because I had both the calling and called sip devices behind a NAT on each end of the call. It may work if only one end is behind a NAT??? -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 5:30 AM Subject: RE: [Asterisk-Users] Bandwdith usage Joseph Shi [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Does anybody know if the voice actually gets routed through Asterisk for calls between SIP devices? I just wonder if calls between SIP devices would take up any bandwidth or CPU at the Asterisk server. Please advise. SIP devices will send re-invitations in an effort to find the most efficient route for the voice data, bypassing the server(s) etc. In a lot of cases, the two endpoints will end up speaking to one another directly. You can set up Asterisk to keep itself in the loop (canreinvite = no), or it might want to remain in the loop regardless of your settings. For instance, Asterisk will want to remain in the loop if you're recording the call - for obvious reasons. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[4]: [Asterisk-Users] ACT Gateways
On Mon, 25 Oct 2004 13:48:35 +0200, Miroslav Nachev [EMAIL PROTECTED] wrote: Yes, I now all prices of Yoda, and I am looking for some partners with which to combine one order for samples. I am only willing to pay for a sample if they - GUARANTEE that their product will work 100% in any possible situation you could encounter in Japan, that is it has to work with NTT analog PSTN lines; analog phone lines delivered by ADSL modem based VoIP services such as YahooBB, OCN, etc; analog ports on all the major ISDN TAs in use over here such as NEC and Yamaha - will refund not only the purchase price but also the shipping charges, tax, duties and wasted time if it turns out that the product does not stack up to the Japanese environment - have at the very least applied for Japanese type approval with JATE If those conditions are not met, then that means that we will end up doing research for the vendor and if we are to do that free of charge, then they must deliver a free sample for testing. Quid pro quo. However, some vendors are smart enough to realise that somebody like us testing their device in Japan will actually add more value than they are to gain from a quick one time sale. The problem is that they require min. 10 pcs per order. I have sent an email to Yoda and I will call them as soon as the current Typhoon has passed over Taiwan and people there will have returned to work. At the moment everything is shut down there. So, let's see how they respond. If they meet the conditions, I am happy to join in on an order. If they don't but provide a sample for testing and the device turns out to be usable here, then, too, I will be happy to join in on an order. The price for samples of VG400 is $320. The device use TI DSP for coding. The prices for regual (big) quantities is $60 per port seems reasonable and affordable. let's see how Yoda respond. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto get voicemail $VM_ vars into externnotify script?
On Mon, 2004-10-25 at 11:09, Wilson Pickett wrote:
I am trying to slap together a script that will email2sms the details of
the voicemails left on my * box to my gsm phone. I can't figure out how
to get my script to pick up the voicemail vars like ${VM_MSGNUM},
${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}.
change the voicemail.conf to include them in the body using the emailbody as in
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have received a ${VM_DUR}
message from ${VM_CALLERID}. The message was left on ${VM_DATE}.
Hi Wilson,
Thanks for your suggestion. I already have normal voicemail notification
via email working. Just as you pointed out. What I want to do besides
this is use a script configured in externnotify=/home/patrick/myapp.sh
in voicemail.conf to email this same data to a special email address
that will forward that data automatically as an sms message to my gsm
phone. This simple script doesn't even work:
$cat myapp.sh
#!/bin/sh
echo $emailbody
So either I am doing something not right or maybe these vars are not
exported (or whatever the right word is) by app_voicemail and can't be
accessed by other applications.
Any ideas?
Thanks,
Patrick
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[Asterisk-Users] 3com with Asterisk
Hi All Has anyone setup a 3com SIP phone with Asterisk? I cant seem to find a way to input the user info such as username and password for the phone to log on to the server. Can someone help? Thanks Doug Reid Director Stormcorp Network Solutions (Pty) Ltd Tel:+27 11 807 1141 Fax:+27 11 807 3504 Mobile: +27 83 989 0008 E-Mail: [EMAIL PROTECTED] Web:www.stormcorp.co.za --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message, you are hereby notified that you must not disseminate, copy or take any action in reliance on it. If you have received this message in error, please notify Stormcorp Network Solutions, its subsidiaries or associates, immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Stormcorp, its subsidiaries or associates. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom200 VMail (MWI)
Possible work around?
exten = asterisk,1,Goto(context,*0,1)
exten = *0,1,VoiceMailMain(${CALLERIDNUM})
obviously you will want to replace context with the appropriate entry.
Thanks,
Steve
[EMAIL PROTECTED]
- Original Message -
From: Arsen Chaloyan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 1:57 AM
Subject: [Asterisk-Users] Snom200 VMail (MWI)
Hi all.
I'm using snom200-SIP 3.54.
I successfully configure snom and asterisk to work
together (thanks to wiki).
But I want snom to send '*0' instead of 'asterisk'
when I press VMail button.
exten = asterisk,1,VoiceMailMain(${CALLERIDNUM})
exten = *0,1,VoiceMailMain(${CALLERIDNUM})
Is it possible to configure snom for this?
Thanks in advance,
Arsen.
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Re: [Asterisk-Users] KSS/BLF on Asterisk
There is also a SIP flash for the 2102 PE phones. I am about to experiment with it as soon as I have some time. Thanks, Steve [EMAIL PROTECTED] - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, October 25, 2004 2:36 AM Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk [EMAIL PROTECTED] wrote: On Mon, 2004-10-25 at 14:05, Paul Hales wrote: First you need to set up the hint function in extensions.conf: exten = 6003,hint,SIP/6003 Then set the SNOM's buttons (FUNCTION KEYS) to be speed dials (DESTINATION) to the extensions in question. Has anyone managed to do this with a polycom IP phone? eg, the IP500/IP600 phones? And I'm starting to wonder if 3com's phones need another look. They claim SIP compliance, and their 3100-series sets look very interesting (aspecially the 3105 attendant console). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip.conf user with defaultip= .... works but callerid not settable (= ip)
Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test123 [1234567]context=outbound type=friendhost=192.168.0.1 insecure=verydefaultip=192.168.0.1username=1234567 callerid="1234567" 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] protection
More on the scenario. Public internet access? Usage from the LAN? Is your machine behind a NAT? Defaultip does nothing as far as protection as far as I know. You can use inkeys and RSA auth or also statements similar to [someiaxphone] type=friend host=dynamic secret=moofoo context=totalaccess notransfer=yes ;deny=0.0.0.0/0.0.0.0 ;permit=68.32.52.90/255.255.255.255 ;qualify=300 Efficient use of access-lists on layer 3 devices can mitigate most threats. Thanks, Steve [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 5:47 AM Subject: [Asterisk-Users] protection hi, how do u prevent unauthorized usage or block users temporarily to use Asterisk services ? Is defaultip and secret enought ? what u do to prevent this. tia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AST doesn't start after update from 0.5 to 1.0
Did you upgrade libpri zaptel and asterisk? - Original Message - From: Joerg Beck [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 6:03 AM Subject: [Asterisk-Users] AST doesn't start after update from 0.5 to 1.0 Please be graceful if I might sound foolish, it's my first posting here. I was happy to have asterisk 0.5 running in a very limited one-phone test configuration. Then I downloaded 1.0 and installed it. Afterwards, asterisk didn't even start any more. The messages are: Notice: iax2-provision.c:496 iax_provision_reload: no iax provisioning configuration found, IAX provisioning disabled. Warning: phx.c:2304 ast_register'_application: Already hvae an application 'Voicemail2' Warning: phx.c:2304 ast_register'_application: Already hvae an application 'VoicemailMain2' Warning: loader.c:334 ast_load_resource: app_voicemail.so: load_module failed, returning -1 Warning: loader.c:429 load_modules: loadin module app_voicemail.so failed! Any help is highly appreciated ! Thank you, Joe. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy setup
I would not trust Linksys to work with dyndns.org without doing some testing. http://www.dyndns.org/news/releases/archives/2003/11/288.html Thanks, Steve [EMAIL PROTECTED] - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Wilson Pickett' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, October 25, 2004 6:45 AM Subject: RE: [Asterisk-Users] IAXy setup [EMAIL PROTECTED] wrote: I haven´t tougth about being re-provisioning the iaxy box :)... But how do you detect the dns change? wich ddns company are u using? Take look at dyndns.org, they have both free and paid services. I set a cron job to run every 10 minutes or so and based on a script I found at DDNS, it detects its ip address and if changed does a bunch of things like notify DDNS so the DNS is updated (some Linksys and other NAT routers will do this btw) and change the ip in the sip.conf, do a sip reload, reprovision the IAXy. works like a charm! I'm not getting how the IAXy obtains the new information. Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [Asterisk-Users] IAX wireless problem
On 23 Oct 2004, at 18:03, Benjamin on Asterisk Mailing Lists wrote: On Sat, 23 Oct 2004 17:10:40 +0200, Neal Nelson [EMAIL PROTECTED] wrote: I'm using IAXComm on the Mac to connect to my Asterisk system and it all seems to work well when I'm connected to my wired network. When I use wireless instead, IAXComm never registers with Asterisk and when I call, ASterisk seems to think it's connected but no sound comes back. did you define your client as host=dynamic in iax.conf? use iax2 debug on the asterisk console to get a session transcript when you try to register and make test calls. if there are no Rx-Frame messages coming in from the client, then you have some sort of connectivity problem with your wireless setup. Use tcpdump or ethereal to see if any traffic is coming in on port 4569. I've got host=dynamic defined in iax.conf. I did have a deaultip entry as well but it seems to make no difference. For a bit more info, I'm running Asterisk 1.0.1 on FreeBSD5.3 and I'm using IPSec to encrypt my wireless connection. I've tried it with a clear wireless connection and it still has the problem. When I use a wired connection I get the following IAX messages: Rx: REGREQ with username Tx: REGAUTH with challenge Rx: REGREQ with response Tx: REGACK Rx: ACK After this I'm all registered and all works well. With wireless I get the following: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 3ms SCall: 1 DCall: 18892 [10.0.1.3:4569] AUTHMETHODS : 2 CHALLENGE : x USERNAME: xx Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 3ms SCall: 1 DCall: 18892 [10.0.1.3:4569] AUTHMETHODS : 2 CHALLENGE : x USERNAME: xx Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 3ms SCall: 1 DCall: 18892 [10.0.1.3:4569] AUTHMETHODS : 2 CHALLENGE : x USERNAME: xx Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 18892 DCall: 0 [10.0.1.3:4569] USERNAME: xx REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 20015ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 20018ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 20015ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 20018ms SCall: 1 DCall: 18892 [10.0.1.3:4569] Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 3ms SCall: 1 DCall: 18892 [10.0.1.3:4569] AUTHMETHODS : 2
Re: Fwd: [Asterisk-Users] IAX wireless problem
On Mon, 25 Oct 2004, Neal Nelson wrote: With wireless I get the following: Your Mac can't hear Asterisk's replies. NAT issue? Firewall? some confusion with a multihomed box? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip.conf user with defaultip= .... works butcallerid not settable (= ip)
try is with just "callerid=1234567" - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test123 [1234567]context=outbound type=friendhost=192.168.0.1 insecure=verydefaultip=192.168.0.1username=1234567 callerid="1234567" 1234567 ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AST doesn't start after update from 0.5 to 1.0
Joerg Beck wrote: [CUT] Warning: phx.c:2304 ast_register'_application: Already hvae an application 'Voicemail2' Warning: phx.c:2304 ast_register'_application: Already hvae an application 'VoicemailMain2' Warning: loader.c:334 ast_load_resource: app_voicemail.so: load_module failed, returning -1 Warning: loader.c:429 load_modules: loadin module app_voicemail.so failed! My guess is that you did not clear out /usr/lib/asterisk/modules and that you have some old modules in there playing tricks with you. Check the date on the xxx.so files. Optionally you can noload = xxx the old modules in /etc/asterisk/modules.conf if you do not want to delete them. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf user with defaultip= .... worksbutcallerid not settable (= ip)
No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip.conf user with defaultip= ....worksbutcallerid not settable (= ip)
I trust you are stopping and restarting asterisk between changes? Try commenting out the defaultip line next. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:06 AM Subject: RE: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 8:17 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | A flex grow is like a channel bank. A normal PRI comes into a router. The | router breaks out some channels for data and the other voice channels become | analog POTS lines. You will need POTS cards. | | I am positive that you could have your T100P and asterisk provide this | function so that you wouldnt need their equipment or POTS. Just depends on | the tech you get whether they will help or not. Just read the first | paragraph of the product description on Digium's site. | http://www.digium.com/index.php?menu=wildcard_t100p | we have sprint with verizon local loop, and sprint cannot see the t100p card but they show the link as being up the decision to send the t100p back to digium or not will be made this morning after fooling with the card for about a month now, without success. calling the sangoma folks this morning since they support all aspects of their cards (data and voice) where digium does not. and they also appear to work with asterisk Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy setup
Steve Totaro said: I would not trust Linksys to work with dyndns.org without doing some testing. http://www.dyndns.org/news/releases/archives/2003/11/288.html If you read down in that article you'll see that is it was written prior to December 15th, 2003. I'd expect most Linksys routers that indicate support for DDNS service from DynDNS.org have been updated by now. My WRV54G didn't originally but now works and has been in service for about 6 months IIRC. $0.02, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf user with defaultip=....worksbutcallerid not settable (= ip)
I just do a -- asterisk -rx reload This picks up the changes in sip.conf for sure :-) If I comment out the defaultip line then asterisk still uses the IP address as callerid :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip=worksbutcallerid not settable (= ip) I trust you are stopping and restarting asterisk between changes? Try commenting out the defaultip line next. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:06 AM Subject: RE: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Cirelle Enterprises wrote: - Original Message - From: Daniel Daley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 21, 2004 5:49 PM Subject: [Asterisk-Users] Digium Wildcard T1 Compatibility | Hi, | | I have a quick question about the T100P. I've used the card before in a | PRI setup and it worked great. I'm now trying to figure out a setup for | another company that gets services from Verizon. They offer what they | call a flexgrow T1 where they say the voice lines are delivered as just | standard POTS channels. Will the wildcard handle this kind of T1 or is | that something you would need to break out into separate lines and go | into POTS cards? | | Thanks, | | --Dan-- | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) RJ48C is *not* standard ethernet cable. The twisted pairs are grouped differently. Ethernet cables work OK for T1s if they are only 2 or 3 metres long. Long ethernet cables give high error rates when used for T1s. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Cirelle Enterprises wrote: - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 8:17 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | A flex grow is like a channel bank. A normal PRI comes into a router. The | router breaks out some channels for data and the other voice channels become | analog POTS lines. You will need POTS cards. | | I am positive that you could have your T100P and asterisk provide this | function so that you wouldnt need their equipment or POTS. Just depends on | the tech you get whether they will help or not. Just read the first | paragraph of the product description on Digium's site. | http://www.digium.com/index.php?menu=wildcard_t100p | we have sprint with verizon local loop, and sprint cannot see the t100p card but they show the link as being up They see the link is up, but can't see the card? What teh heck is that supposed to mean? Its sounds suspiciously like telco drivel. :-) the decision to send the t100p back to digium or not will be made this morning after fooling with the card for about a month now, without success. calling the sangoma folks this morning since they support all aspects of their cards (data and voice) where digium does not. and they also appear to work with asterisk Since 90% of problms are due to clueless telco people who don't know how to get their end right, what makes you think a Sangoma card will help? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hi, I am using open323 version: 1.13.5 pwlib verison : 1.6.6 OH323 version: 0.6.3b Can this be a configuration problem? Here is my config data: [general] listenAddress=xxx.xxx.xxx.xxx listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=yes h245Tunnelling=no h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=3 libTraceFile=tr.out gatekeeper=DISABLE userInputMode=TONE amaFlags=default accountCode=H323 context=htest Thanks, Ender Erbey conacom GmbH Vlasis Hatzistavrou wrote: Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Out of curiousity, would you mind sharing what you have tried? - Original Message - From: Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:11 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 8:17 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | A flex grow is like a channel bank. A normal PRI comes into a router. The | router breaks out some channels for data and the other voice channels become | analog POTS lines. You will need POTS cards. | | I am positive that you could have your T100P and asterisk provide this | function so that you wouldnt need their equipment or POTS. Just depends on | the tech you get whether they will help or not. Just read the first | paragraph of the product description on Digium's site. | http://www.digium.com/index.php?menu=wildcard_t100p | we have sprint with verizon local loop, and sprint cannot see the t100p card but they show the link as being up the decision to send the t100p back to digium or not will be made this morning after fooling with the card for about a month now, without success. calling the sangoma folks this morning since they support all aspects of their cards (data and voice) where digium does not. and they also appear to work with asterisk Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip.conf user withdefaultip=....worksbutcallerid not settable (= ip)
What type of phone? What setting on phone. Try commenting out the host line. Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:25 AM Subject: RE: [Asterisk-Users] sip.conf user withdefaultip=worksbutcallerid not settable (= ip) I just do a -- asterisk -rx reload This picks up the changes in sip.conf for sure :-) If I comment out the defaultip line then asterisk still uses the IP address as callerid :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip=worksbutcallerid not settable (= ip) I trust you are stopping and restarting asterisk between changes? Try commenting out the defaultip line next. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:06 AM Subject: RE: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Wildcard T1 Compatibility (ethernet f or T1 cables)
snip RJ48C is *not* standard ethernet cable. The twisted pairs are grouped differently. Ethernet cables work OK for T1s if they are only 2 or 3 metres long. Long ethernet cables give high error rates when used for T1s. Regards, Steve snip I have run several robbed-bit(non-PRI) T1s on standard CAT5e ethernet cable for runs of over 100 feet(30 meters) and have seen no increase in the error rate of the T1s. I've done this for years, and even the telco install techs say it really doesn't matter especially if you are running robbed-bit, supposedly its more forgiving than PRIs are. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf user with defaultip=....worksbutcallerid not settable (= ip)
PS.. If I send NO callerid at all from my cisco, asterisk translates the callerid to the ip address where the call originates from (in this case my cisco).. When I DO send a callerid from my cisco it uses THAT callerid But still I don't get it to overrule the callerid by setting the callerid= line -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip=worksbutcallerid not settable (= ip) I trust you are stopping and restarting asterisk between changes? Try commenting out the defaultip line next. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:06 AM Subject: RE: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:26 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility Cirelle Enterprises wrote: - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 8:17 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | A flex grow is like a channel bank. A normal PRI comes into a router. The | router breaks out some channels for data and the other voice channels become | analog POTS lines. You will need POTS cards. | | I am positive that you could have your T100P and asterisk provide this | function so that you wouldnt need their equipment or POTS. Just depends on | the tech you get whether they will help or not. Just read the first | paragraph of the product description on Digium's site. | http://www.digium.com/index.php?menu=wildcard_t100p | we have sprint with verizon local loop, and sprint cannot see the t100p card but they show the link as being up They see the link is up, but can't see the card? What teh heck is that supposed to mean? Its sounds suspiciously like telco drivel. :-) the decision to send the t100p back to digium or not will be made this morning after fooling with the card for about a month now, without success. calling the sangoma folks this morning since they support all aspects of their cards (data and voice) where digium does not. and they also appear to work with asterisk Since 90% of problms are due to clueless telco people who don't know how to get their end right, what makes you think a Sangoma card will help? Steve agreed! did they even send out a tech with a tbird? Have you attach a loopback? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help for Newbie?
Hello all. I am new to the list and after doing some research on Asterisk this week I would like to get started testing. I am a 15 year Unix veteran and Open Source User. I was wondering what you guys would suggest I use to start testing with regards to the Telephone Interface? I am the IT Manager for a firm that is moving next year. I have decided to go with VoIP for our new Phone System. After discovering Asterisk last week, I have decided to test it and get some practical experience using the software. I currently have an unused ISDN (BRI) line that I was thinking about cancelling until I learned of Asterisk. I thought about buying one of the BRI PCI cards (listed on the Digium website) to use in a test server. Although, I see that they can be rather expensive for something that I most likely will have to just throw away when we move (I assume we will have a T1 or fractional T1 in the new building.). My question is, What would you guys recommend I use to get started testing/looking at this software? I only see these two (affordable) options for testing: Buy ISDN BRI interface board, use to test and then throw it away. (This would give me two lines to test with, which would be nice.) Buy an analog board and use a dedicated line. (I think we only have one incoming line that I could use for testing. OR, could I use the two POTS ports from our ISDN router and in effect just use the two ISDN data lines as POTS lines?) Thanks to all! Shayne ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf user withdefaultip=....worksbutcalleridnot settable (= ip)
My UA is not a phone it's a cisco AS5350 gateway When I comment out the host= line calling doesn't work anymore (asterisk uses the default context then which doesn't allow calling at all :-) If I set the defaultip= line then (and keep commenting out the host= line) it works again.. But the callerid still doesn't get overruled -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user withdefaultip=worksbutcalleridnot settable (= ip) What type of phone? What setting on phone. Try commenting out the host line. Thanks, Steve Totaro [EMAIL PROTECTED] www.totarotechnologies.com - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:25 AM Subject: RE: [Asterisk-Users] sip.conf user withdefaultip=worksbutcallerid not settable (= ip) I just do a -- asterisk -rx reload This picks up the changes in sip.conf for sure :-) If I comment out the defaultip line then asterisk still uses the IP address as callerid :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip=worksbutcallerid not settable (= ip) I trust you are stopping and restarting asterisk between changes? Try commenting out the defaultip line next. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:06 AM Subject: RE: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) No it's the same... it still uses the IP address as callerid I tried the following ones: callerid=1234567 callerid=1234567 callerid=1234567 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user with defaultip= worksbutcallerid not settable (= ip) try is with just callerid=1234567 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 8:33 AM Subject: [Asterisk-Users] sip.conf user with defaultip= works butcallerid not settable (= ip) Hello I have this in my sip.conf ... it's a cisco which I authenticate on it's IP adress this works.. asterisk authenticates sip calls from this IP as user 1234567 and uses the right context, only, the CALLERID is the ip adress e.g. 19216801 instead of the callerid which i try to set using the 'callerid=' field if I make a user authenticate with his username and password the callerid part does work does anyone know what I am doing wrong? sip.conf: ;Test 123 [1234567] context=outbound type=friend host=192.168.0.1 insecure=very defaultip=192.168.0.1 username=1234567 callerid=1234567 1234567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk-oh323
Hello to all, Im trying to get h323 working with Asterisk, Ive downloaded all require modules (most are .tar.gz files, but if some body knows where to find working rpm file, it will help me), Ive installed Pwlib, Openh323, and Asterisk. When I want to compile the asterisk-oh323 module, I got an error. The error has been reporting to the list, but I did not find the answer. I found another openh323 version (V. 1.13.5) and a patch to this version, but nothing work. Can somebody know how to make Asterisk Work with H323 ?? Which files are needed ?? Where to find Working files ?? I have both Fedora Core 1 and RedHat 9. Thanks. image002.jpg___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Wildcard T1 Compatibility
we have sprint with verizon local loop, and sprint cannot see the t100p card but they show the link as being up the decision to send the t100p back to digium or not will be made this morning after fooling with the card for about a month now, without success. calling the sangoma folks this morning since they support all aspects of their cards (data and voice) where digium does not. and they also appear to work with asterisk Greg ___ I agree with Steve, I have been doing this a long time and 80% of the time if the T1 does not come up it is misconfiguration on the telco end.If Sprint can see the link up but can't see the card, it sounds like verizon still has some piece of equipment in the circuit. Have you tried putting a hard loopback on the circuit and removing it to see if Sprint show the circuit going up and down? I don't believe switching cards is going to help at this point. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage Softphone--outbound calls work, inbound do not
I have successfully installed and configured a HandyTone 286 and can send
and receive calls between the console and the HT adapter. I have also
registered with the Vonage server (via a softphone account) and can place
calls from the HT phone to my cellphone (or any other number) through the
Vonage account. However, I'm unable to call my Vonage softphone number and
have it ring the HT phone. I expect to see debug messages in the Asterisk
console, but I don't see those either. Here are my entire sip.conf and
extensions.conf files. Any assistance you can lend is greatly appreciated.
sip.conf:
[general]
port=5060
;bindaddr=0.0.0.0
bindaddr=192.168.1.104
context=default
register = VonageNumber:VonagePassword@sphone.vopr.vonage.net:5061/
[]
type=friend
username=
secret=password
defaultip=192.168.1.103
host=dynamic
disallow=all
allow=ulaw
allow=alaw
[VonageNumber]
type=friend
username=VonageNumber
secret=VonagePassword
host=sphone.vopr.vonage.net
port=5061
maxexpirey=15
dtmfmode=inband
fromuser=VonageNumber
fromdomain=sphone.vopr.vonage.net
canreinvite=no
nat=no
context=default
disallow=all
allow=all
extensions.conf:
[general]
static=yes
writeprotect=no
[local]
include = default
[default]
exten = s,1,Answer
;exten = s,2,VoicemailMain
exten = ,1,Dial(SIP/)
exten = ,2,Hangup
exten = ,1,Answer
exten = ,2,VoicemailMain
exten = VonageNumber,1,Answer
exten = VonageNumber,2,Dial(SIP/)
exten = _9.,1,Dial(SIP/${EXTEN:1}@VonageNumber,30,r)
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RE: [Asterisk-Users] Help for Newbie?
Hi Shayne, What about starting with a single line analog pots card, this way you can take it home and use it there with your tests to get up to speed on what works and what doesnt. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help for Newbie? Hello all. I am new to the list and after doing some research on Asterisk this week I would like to get started testing. I am a 15 year Unix veteran and Open Source User. I was wondering what you guys would suggest I use to start testing with regards to the Telephone Interface? I am the IT Manager for a firm that is moving next year. I have decided to go with VoIP for our new Phone System. After discovering Asterisk last week, I have decided to test it and get some practical experience using the software. I currently have an unused ISDN (BRI) line that I was thinking about cancelling until I learned of Asterisk. I thought about buying one of the BRI PCI cards (listed on the Digium website) to use in a test server. Although, I see that they can be rather expensive for something that I most likely will have to just throw away when we move (I assume we will have a T1 or fractional T1 in the new building.). My question is, What would you guys recommend I use to get started testing/looking at this software? I only see these two (affordable) options for testing: Buy ISDN BRI interface board, use to test and then throw it away. (This would give me two lines to test with, which would be nice.) Buy an analog board and use a dedicated line. (I think we only have one incoming line that I could use for testing. OR, could I use the two POTS ports from our ISDN router and in effect just use the two ISDN data lines as POTS lines?) Thanks to all! Shayne ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hello, As a start, you can change h245Tunnelling to yes. This will probably solve the problem, as I would receive the messages that you described when I had problems with the H245 negotiation. In addition what are your [codecs] settings in oh323.conf? I assume that you use G729A on the A800 as is shown on the diagram, but what about the settings that you have on the Asterisk side? Also, which version of Asterisk are you using? Best regards, Vlasis. Ender Erbey wrote: Hi, I am using open323 version: 1.13.5 pwlib verison : 1.6.6 OH323 version: 0.6.3b Can this be a configuration problem? Here is my config data: [general] listenAddress=xxx.xxx.xxx.xxx listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=yes h245Tunnelling=no h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=3 libTraceFile=tr.out gatekeeper=DISABLE userInputMode=TONE amaFlags=default accountCode=H323 context=htest Thanks, Ender Erbey conacom GmbH Vlasis Hatzistavrou wrote: Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility (ethernet f or T1 cables)
mattf wrote: snip RJ48C is *not* standard ethernet cable. The twisted pairs are grouped differently. Ethernet cables work OK for T1s if they are only 2 or 3 metres long. Long ethernet cables give high error rates when used for T1s. Regards, Steve snip I have run several robbed-bit(non-PRI) T1s on standard CAT5e ethernet cable for runs of over 100 feet(30 meters) and have seen no increase in the error rate of the T1s. I've done this for years, and even the telco install techs say it really doesn't matter especially if you are running robbed-bit, supposedly its more forgiving than PRIs are. MATT--- Your mileage many vary, as they say. I've had problems on T1 and E1 cables as short as 10 metres. Changing the pairing stopped the errors completely. You said you can see no *rise* in the error rate. If the error rate is even measurable, I'd say it is too high. I can usually leave a link for weeks, and it records zero errors. Robbed bit is more forgiving for dropped calls, as sensible terminals persistence check the signaling. The audio suffers just as many clicks and pops, though. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:21 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) | | | RJ48C is *not* standard ethernet cable. The twisted pairs are grouped | differently. Ethernet cables work OK for T1s if they are only 2 or 3 | metres long. Long ethernet cables give high error rates when used for T1s. | According to the t1 techs I've been dealing with, a standard ethernet cable will work (we have been using a 3 foot segment approx 1m) In fact the only configuration that lights all lights is a standard ether cable. T1 cross-over doesn't create a proper link and lights on the t100p and smart jack indicate a failed condition. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:41 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | agreed! did they even send out a tech with a tbird? Have you attach a | loopback? The line is in good shape and works with a cisco 2620 inline. when we unplug the cisco and plug in the t100p, (using the same cable - the end comes out of the cisco wic and is plugged directly into the t100p) the link remains up, but telco cannot see the card. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Cirelle Enterprises wrote: - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:41 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | agreed! did they even send out a tech with a tbird? Have you attach a | loopback? The line is in good shape and works with a cisco 2620 inline. when we unplug the cisco and plug in the t100p, (using the same cable - the end comes out of the cisco wic and is plugged directly into the t100p) the link remains up, but telco cannot see the card. What signaling are you using, and what is in your zapata.conf file? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Cirelle Enterprises wrote: - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:21 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) | | | RJ48C is *not* standard ethernet cable. The twisted pairs are grouped | differently. Ethernet cables work OK for T1s if they are only 2 or 3 | metres long. Long ethernet cables give high error rates when used for T1s. | According to the t1 techs I've been dealing with, a standard ethernet cable will work (we have been using a 3 foot segment approx 1m) In fact the only For such a short cable it will be OK. configuration that lights all lights is a standard ether cable. T1 cross-over doesn't create a proper link and lights on the t100p and smart jack indicate a failed condition. Surprise. Surprise. If you cross all the wires over. it doesn't work. :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:32 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | Out of curiousity, would you mind sharing what you have tried? | from a previous post... T1 provider info HDLC 24 channel 160.81.118.46/30 (my side) 160.81.118.45/30 (sprint) # ifconfig hdlc0 Link encap:(Cisco)-HDLC inet addr:160.81.118.46 P-t-P:160.81.118.45 Mask:255.255.255.252 UP POINTOPOINT RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:1399 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:50 RX bytes:0 (0.0 b) TX bytes:40388 (39.4 Kb) loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:3468 errors:0 dropped:0 overruns:0 frame:0 TX packets:3468 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:141932 (138.6 Kb) TX bytes:141932 (138.6 Kb) commands to achieve the above: /sbin/modprobe zaptel /sbin/modprobe wct1xxp /sbin/modprobe hdlc /sbin/ztcfg /sbin/sethdlc hdlc0 cisco /sbin/ifconfig hdlc0 arp multicast 160.81.118.46 pointopoint 160.81.118.45 netmask 255.255.255.252 /sbin/route add default gw 160.81.118.45 netmask 0.0.0.0 dev hdlc0 /proc/sys/net/ipv4/ip_forward = 1 sysctl.conf: # Controls IP packet forwarding net.ipv4.ip_forward = 1 /etc/sysconfig/network: NETWORKING=yes HOSTNAME=ast.cirelle.com IPV4_FORWARD=yes GATEWAY= iptables, accepts in all directions (no firewalling) zaptel.conf: span=1,1,5,esf,b8zs nethdlc=1-24 loadzone = us defaultzone=us zapata.conf switchtype = national signalling = pri_cpe loadzone=us defaultzone=us ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip.conf user withdefaultip=....worksbutcallerid not settable (= ip)
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:43 AM Subject: RE: [Asterisk-Users] sip.conf user withdefaultip=worksbutcallerid not settable (= ip) PS.. If I send NO callerid at all from my cisco, asterisk translates the callerid to the ip address where the call originates from (in this case my cisco).. When I DO send a callerid from my cisco it uses THAT callerid Problem solved then. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
- Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 10:17 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | configuration that lights all lights is a standard ether cable. T1 cross-over | doesn't create a proper link and lights on the t100p and smart jack indicate | a failed condition. | | | Surprise. Surprise. If you cross all the wires over. it doesn't work. :-) | | Steve T1 cross-over was suggested by digium tech support Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
Cirelle Enterprises wrote: - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:32 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | Out of curiousity, would you mind sharing what you have tried? | from a previous post... T1 provider info HDLC 24 channel 160.81.118.46/30 (my side) 160.81.118.45/30 (sprint) # ifconfig hdlc0 Link encap:(Cisco)-HDLC inet addr:160.81.118.46 P-t-P:160.81.118.45 Mask:255.255.255.252 UP POINTOPOINT RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:1399 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:50 RX bytes:0 (0.0 b) TX bytes:40388 (39.4 Kb) loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:3468 errors:0 dropped:0 overruns:0 frame:0 TX packets:3468 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:141932 (138.6 Kb) TX bytes:141932 (138.6 Kb) commands to achieve the above: /sbin/modprobe zaptel /sbin/modprobe wct1xxp /sbin/modprobe hdlc /sbin/ztcfg /sbin/sethdlc hdlc0 cisco /sbin/ifconfig hdlc0 arp multicast 160.81.118.46 pointopoint 160.81.118.45 netmask 255.255.255.252 /sbin/route add default gw 160.81.118.45 netmask 0.0.0.0 dev hdlc0 /proc/sys/net/ipv4/ip_forward = 1 sysctl.conf: # Controls IP packet forwarding net.ipv4.ip_forward = 1 /etc/sysconfig/network: NETWORKING=yes HOSTNAME=ast.cirelle.com IPV4_FORWARD=yes GATEWAY= iptables, accepts in all directions (no firewalling) zaptel.conf: span=1,1,5,esf,b8zs nethdlc=1-24 loadzone = us defaultzone=us zapata.conf switchtype = national signalling = pri_cpe loadzone=us defaultzone=us If you want to use the whole T1 for HDLC, why have you configured it as an ISDN PRI line? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
On Mon, 25 Oct 2004, Cirelle Enterprises wrote: - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:21 AM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility | | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) | | RJ48C is *not* standard ethernet cable. The twisted pairs are grouped | differently. Ethernet cables work OK for T1s if they are only 2 or 3 | metres long. Long ethernet cables give high error rates when used for T1s. According to the t1 techs I've been dealing with, a standard ethernet cable will work (we have been using a 3 foot segment approx 1m) In fact the only configuration that lights all lights is a standard ether cable. T1 cross-over doesn't create a proper link and lights on the t100p and smart jack indicate a failed condition. The pairs in an ethernet cable are the same pairs as a t1 cable, they are paired 12, 36, 45, 78. T1 uses 12 and 45, ethernet uses 12, 36. If you have a good cable it should work fine. Going too long of a distance from your t1 mounting may required a higher LBO on each end to compensate for the additional copper. Also make sure you are using 24 awg wire, not the 26 or 28 awg in most patch cables. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zultys Zip 2 Setup
It was trial and error for us, too. Here's a config that works for us with *. Zultys is only politely interested in supporting their phones with non-Zultys systems: ROMAVERSION 3.52 IF0DHCP DHCP SERVERIP 216.xxx.xxx.xx SERVERPORT 5060 DOMAINNAME wpti.net SERVERREGISTER YES DIALPLAN 9|7xx|50xx|0xxx|*x.|1xx|xxx TRANSPORT_TYPE UDP LINE1PORT 5060 LINE1AEC YES SIP_MESSAGE_WAITING YES SIP_SEND_PRACK NO SIP_URI_USER_PARAM NO OOBTELEVENTS OOB_RFC2833 TELEVENTPAYLOAD 101 DROPVOICE YES SQUELCHDTMF NO ABCDMODE TRANSITION G711UON YES G711UPACK 20 G711USS NO G711AON YES G711APACK 20 G711ASS NO G729ON YES G729PACK 20 G729SS NO AJB_MAXDELAY 100 FJB_DELAY 40 AUTO_JB_SWITCH NO COUNTRY USA NTPSERVERIP 192.43.244.18 TIMEZONE -420 DST YES RINGTONE 1 LINE1NUMBER 90055522368 LINE1AUTHUSER 9005552368 LINE1AUTHPSWD pw3268 LINE1CALLERID John Public 900-555-2368 Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 23 Oct 2004, Me wrote: I bought one of these phones and I am trying to set it up. So far, I have figured out how to get to the web interface but I can't seem to figure out how to set some of the most important things like the Proxy address etc.. The manual is useless for things like this as well as their website. The only thing these folks seem to give instructions on is how to change the volume etc, but nothing related to actually setting up the phone for use with asterisk or anything else. The Uniden phone was pretty much the same thing, virtually zero docs on how to get started etc.. So far the cheapest phone (the GrandStream) has been the most straight forward to setup. I have already boxed up the Uniden which is ashame since it's a great phone. Thing is I can't use it behind a NAT so it has to go back :( I did email them though and ask them if they had the new firmware ready.. -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-23%5C7c19079bf3e44d948f0e40a70fab4469C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip CallerPres support?
Roy et All, If someone could expand on CallerPres requirements in chan_sip I can do the work. I have added numerous extras to chan_sip already, RADIUS, new CDRs, Dynamic Dial plans, Find-Me, Follow-Me and such. I am just one programmer, but let me know what needs to be done and I can create the code fairly quickly. Race Vanderdecken Asterisk aT vanderDecken period coM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: 24 October 2004 08:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip CallerPres support? hi would it be hard to implement CallerPres support in chan_sip? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Wildcard T1 Compatibility
On Mon, 25 Oct 2004, Henry Devito wrote: I agree with Steve, I have been doing this a long time and 80% of the time if the T1 does not come up it is misconfiguration on the telco end.If Sprint can see the link up but can't see the card, it sounds like verizon still has some piece of equipment in the circuit. Have you tried putting a hard loopback on the circuit and removing it to see if Sprint show the circuit going up and down? I don't believe switching cards is going to help at this point. Need to be devious. Can you see the loop now? says he innocently, as he accidently forgets to loop the circuit. Oh yes, they say. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] Digium TheVoice recordings' sound
Email her!!! She is awesome and will of course fix them for you. She doesn't work for digium. Most of the sound files that are included with Asterisk were paid for by various companies and released under GPL so that she would still be able to make a living, and the entire community would benefit. Again, email her... she will make it right, ~c -- Message: 6 Date: Sun, 24 Oct 2004 12:49:46 -0500 From: Kristian Kielhofner [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible To: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Benjamin on Asterisk Mailing Lists wrote: On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote: http://www.theivrvoice.com/ would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. I think when he wrote 'She does work at Digium' it was meant in the sense of She is doing work at Digium', or 'She does (some) work for Digium ;-) rgds benjk I would lean towards she does some work for Digium. Did you check out her webpage? One, she lives in Canada, so she certainly does not work at Digium in the physical sense. Two, her client list leads me to believe that Digium is probably one of her smaller clients. Did you try to contact her directly? She seems to imply on her site that customer satisfaction is pretty important to her. Maybe she will fix them for you. You did after all pay for these files, right? ;) She must have a fairly good relationship with Digium, however, because she does have the official title of Asterisk Diva, and she was at Astricon. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cheap hosted servers and Asterisk
We have several servers running that way. Virtual Machines, they seem to be running fine . XMC - Your VOIP Solutions and Consulting Experts. Ask us about low cost Asterisk PBX and VOIP legacy gateways. Ask us for the best prices in T1's for data and voice services nationwide. We will not be beat! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: Saturday, October 23, 2004 10:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cheap hosted servers and Asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Saturday, October 23, 2004 12:37 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Cheap hosted servers and Asterisk Does anyone have any experience with running Asterisk on dedicated servers from any of the cheap hosting providers, like 11? I'd like to get my asterisk/mail/web server out of my house. There isn't a whole lot of traffic involved, but I'd rather not end up with someplace that *utterly* oversubscribes their bandwidth--it needs to work with Asterisk, not just TCP-based services. I can find a number Haven't had any experience, however if your clients connecting to Asterisk are setup to properly reinvite (assume you are using sip) then you shouldn't have large overage charges. If your using Asterisk in the media path, the potential for overage charges then increases. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNID in chan_sip.c
Hey list, I have now been looking at asterisk for a few weeks, trying to solve a particular problem I have. Let me elaborate. ++ | !asterisk | SIP trunk+---+ | Softswitch |---| asterisk | || class 4 peer +---+ ++ | | | | users in voip/pstn The Softswitch takes care of normal call routing, asterisk is to be used solely for voicemail and for IVR. So users on the softswitch will have to set up CF to the magic voicemail number, which is routed to a SIP trunk leading to the asterisk. VM Logic on the asterisk then has to analyse the A number and the B numbers to figure out one of 3 cases: 1. There is a mailbox for A - go right to read my messages 2. There is a mailbox for B - go right to leave message 3. A and B are both unknown, vm wise, authenticate, then go to 1 or reject. In case 1, the B number will most likely be identical to the magic voicemail number, and is therefore irrelevant. In case 2, the A number will be someone unknown, and the magic voicemail number will be the C number. Now, in SIP this is signalled in INVITEs naturally, and a typical invite for a case 2 scenario could look like. INVITE sip:magicvmnum@asterisk-ip:asterisk-port;user=phone SIP/2.0 Allow: UPDATE Call-ID: 6b41f335@softswitch-ip Contact: sip:ANUM@softswitch-ip:softswitch-sipport;user=phone Content-Type: application/sdp CSeq: 1561 INVITE From: sip:ANUM@softswitch-ip;user=phone;tag=6b41f336 Max-Forwards: 31 Reason: sip;cause=302 Require: 100rel To: sip:BNUM@asterisk-ip;user=phone User-Agent: Softswitch Agent Via: SIP/2.0/UDP softswitch-ip:softswitch-sipport;branch=z9hG4bK-2FFF Content-Length: 215 v=0 o=cp10 109870486669 109870486669 IN IP4 softswitch-ip s=SIP Call c=IN IP4 softswitch-ip t=0 0 m=audio some RTP port RTP/AVP 0 8 b=AS:64 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=ptime:30 a=maxptime:30 The 3 SIP adresses that asterisk requires to decode from the INVITE in order for me to handle my extension logic are in the 1st headerline, in the From header and in the To header. Now finally to the problem. In chan_sip.c, there is NO reading of the To: header at all on INVITEs, the information needed to route the call is in the command line (INVITE blabla), but in order for my logic to work, I need the BNUM from the To header. It would be logical to place the address from the To: header in the DNID (which is == EXTEN in these cases), would it not? or at least the user part of the address? I have written a patch for this, but before i post i want to ask you guys if you agree that DNID is the place for it? An alternative would be to place it in username, which is empty in this case aswell, but that would require a new variable possibly called SIPUSERNAME. What do you think? Can stuff get broken by filling stuff into the dnid field of the channel struct from chan_sip.c in this way? As far as I see it, we have to agree that not looking at the To: header at all on INVITEs (not using __get_header anyway) has to be fixed? brgs, Jesper Dalberg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How useful is the screen on IP phones?
João Amaro wrote: I'm using 14 cisco 7940 as Dynamic queue agents. They use the pixel based screen, to login/logout from queues. They can also see the queues stats. Now that's really not fair, to post a message like this without links to the code and/or documentation :-( Can we assume from your message that you have implemented this yourself but are not making it available to the community? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-office topology suggestions
We are looking at putting Asterisk into use at our company. We have pushed it past proof of concept training and would like to roll it out in the very near future. One stumbling block remains: We have five offices in Canada. Our main office is in Edmonton , with branch offices all over the nation. I would like to place the Asterisk server in the Edmonton office and have it route calls to the branch offices. I would also like to have each of the branch offices have a local phone number. That local phone # would actually dial into the Asterisk box , and then routed appropriately via VPN to the correct location. This gives us a method of controlling and tracking all calls made to all offices. The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? I have only tested the Asterisk box with a TDM11B and have no real experience with T1s . Would they help in this situation? Thanks in advance Shawn Dillon PS- My previous post on the issue of my TDM11B is now resolved. It was a dead FXO module. Thanks for the responses. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum A800 OH323 problem
Hi, Thanks a lot! H245 Tunelling solved my problem. Ender Erbey conacom GmbH Vlasis Hatzistavrou wrote: Hello, As a start, you can change h245Tunnelling to yes. This will probably solve the problem, as I would receive the messages that you described when I had problems with the H245 negotiation. In addition what are your [codecs] settings in oh323.conf? I assume that you use G729A on the A800 as is shown on the diagram, but what about the settings that you have on the Asterisk side? Also, which version of Asterisk are you using? Best regards, Vlasis. Ender Erbey wrote: Hi, I am using open323 version: 1.13.5 pwlib verison : 1.6.6 OH323 version: 0.6.3b Can this be a configuration problem? Here is my config data: [general] listenAddress=xxx.xxx.xxx.xxx listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=yes h245Tunnelling=no h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=3 libTraceFile=tr.out gatekeeper=DISABLE userInputMode=TONE amaFlags=default accountCode=H323 context=htest Thanks, Ender Erbey conacom GmbH Vlasis Hatzistavrou wrote: Hello, I am passing traffic between Asterisk and A800's with OH323 without problems. No calls are disconnected after 20 seconds. Which version of Asterisk, OH323, pwlib and openh323 are you running? Best regards, Vlasis. Ender Erbey wrote: Hi, I experienced an interesting problem when i try to make such a connection BRI -- Asterisk (OH323) -- Quintum A800 codec g729a Call begins without problem but it is closed after nearly 20 seconds. Reason : EndedByRemote When i terminate call at cisco gateways with same configuration there is no problem. But with Quintum i have this problem. Have someone an idea about this problem? Thanks, Ender Erbey conacom GmbH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Greetings Ben, If you have not gotten your answer yet; Converting audio files from WAV to telephone 3000HZ will always cause a degradation of the audio quality. Could you elaborate on they sound really bad please? I would take the .gsm files that Digium produced or converted from the .WAV files and play them back on the Windows machine and then compare that quality to the WAV on the Windows box. The .gsm should sound the same on the windows box as on the phone. If the quality is worse then contact Digium. I used http://www.audioi.com/ 'Audio Converter and Ripper' to convert .wav to .gsm. On playback the .gsm has good fidelity. On the phone, we are using a Cisco 7200 and not Asterisk for play back, the quality is the same. Make sure you are converting correctly, 16bit 8000hz Mono, for gsm g729. 1. convert the .wav to .gsm using Audio Converter and Ripper, shareware. 2. play the .gsm and compare to the .wav. 3 play the .gsm and compare to results from the phone. 4. if there is much difference check the phone AUDIO CODECS in asterisk, you might be converting from GSM to G729 on the fly ( I know, technically only a slight difference.) Do you have the G729 license from Digium installed? 5. make sure the conversion process was done correctly. 16bit, 8000hz per sample, Mono. Race Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: 24 October 2004 06:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Digium TheVoice recordings' sound terrible A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? thanks rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-office topology suggestions
Shawn Dillon wrote: The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? I have only tested the Asterisk box with a TDM11B and have no real experience with T1's . Would they help in this situation? The simplest solution is to buy phone numbers in those remote cities from a VOIP provider, rather than a PSTN provider. That provider can then deliver the inbound calls to your Asterisk server using a pure IP connection. Check the Wiki for VOIP providers in Canada; I'm sure there are a number who can do what you need. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How useful is the screen on IP phones?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: | João Amaro wrote: | | I'm using 14 cisco 7940 as Dynamic queue agents. | | They use the pixel based screen, to login/logout from queues. | They can also see the queues stats. | | | Now that's really not fair, to post a message like this without | links to the code and/or documentation :-( | | Can we assume from your message that you have implemented this | yourself but are not making it available to the community? | | Hi It's not finished yet. I've to make some changes, because right now i've made it to work with my configuration (with just 2 queues). It's based on qview.pl from contrib files in asterisk source. I'll share it :) You can contact me by email . -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBfRgCJUm/Bor63CERAlAwAJ9bzrt619EyrMRFoCpTHNZM2rDdXgCgodxt GZQuS862uSKS/RIzZkKG5Ws= =cIKG -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip.conf user withdefaultip=....worksbutcalleridnot settable (= ip)
No my problem is not solved... Because I still want Asterisk to overrule the callerid :-( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, October 25, 2004 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip.conf user withdefaultip=worksbutcalleridnot settable (= ip) - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 25, 2004 9:43 AM Subject: RE: [Asterisk-Users] sip.conf user withdefaultip=worksbutcallerid not settable (= ip) PS.. If I send NO callerid at all from my cisco, asterisk translates the callerid to the ip address where the call originates from (in this case my cisco).. When I DO send a callerid from my cisco it uses THAT callerid Problem solved then. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-office topology suggestions
On Oct 25, 2004, at 8:09 AM, Kevin P. Fleming wrote: Shawn Dillon wrote: The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? I have only tested the Asterisk box with a TDM11B and have no real experience with T1's . Would they help in this situation? The simplest solution is to buy phone numbers in those remote cities from a VOIP provider, rather than a PSTN provider. That provider can then deliver the inbound calls to your Asterisk server using a pure IP connection. Another alternative would be to buy something like a Sipura 3000 for each branch office and use a local POTS line in each office. The Sipura will then send incoming calls directly to the central Asterisk server. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SayNumber application - in spanish?
After reading the wiki, it would appear that the SetLanguage application will set the proper language variable for use by the rest of the speaking applications, such as SayNumber. However, from my (limited) knowledge of counting in Spanish, it's not quite as straightforward as just having different samples - numbers such as 101 don't follow the one hundred+one format of English... This being said, do the numeral-speaking applications within asterisk understand these syntactic differences in languages, or am I on my own for creating a Spanish IVR? -j, who is not looking forward to reimplementing SayNumber() for Spanish in perl... -- Jeffrey Paul Senior Network Administrator - Group Financial LLC 248.331.1970 - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How useful is the screen on IP phones?
Kevin P. Fleming wrote: João Amaro wrote: I'm using 14 cisco 7940 as Dynamic queue agents. They use the pixel based screen, to login/logout from queues. They can also see the queues stats. Now that's really not fair, to post a message like this without links to the code and/or documentation :-( Can we assume from your message that you have implemented this yourself but are not making it available to the community? Probably not using the SIP image too ;) Because with SIP you can only GET pages not PUSHing them to the user :) But the other Cisco images provide some usefull coding abilities, but then again, is it more usefull then a PC with an integrated softclient (in the target software) + headset? I personaly don't think so. Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage Softphone--outbound calls work, inbound do not
El 25/10/2004, a las 15:53, Richard Branham escribió: register = VonageNumber:VonagePassword@sphone.vopr.vonage.net:5061/ Maybe your incoming calls are going to a non existent number in your system ??? try register = VonageNumber:VonagePassword@sphone.vopr.vonage.net:5061/ ·· Adrià Vidal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How useful is the screen on IP phones?
Thanks for the reply -- how well is this documented? Is information available from Cisco to endusers, or is this a big-money affair only? -Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: Friday, October 22, 2004 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How useful is the screen on IP phones? Hey Jay All the stuff you've described is possible. I've done some playing with the XML services on a Cisco 7960 to give ACD queue stats and system uptime info. The phone has a mini web browser built in so it's pretty easy to knock up some glue scripts in the back end to do what you want to do. I think there are some examples on the wiki too. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
