[Asterisk-Users] email notification when leaving a message
Hi! I have configured: iax.conf; voicemail.conf extensions.conf everything works fine... the only things.. i do not receive any email notification when a voicemail is left on the *.. any clues??? i think my email server works(?).. In fact i am able to send an email to the root (mail root etc...).. but aside that.. i am not able to send any other email outside the * box... any clues on how to solve that... I have installed * on RedHat.. Thanks for your help, AlexC __ Lèche-vitrine ou lèche-écran ? magasinage.yahoo.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
On Mon, 2005-05-02 at 16:54 -0400, Patrick M. Gray, Jr. wrote: > No errors, asterisk just immediately sends the other call to voicemail if > there > is already a call in progress. Try turning on "Call waiting" on your cisco phone. Cheers, Joris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Pedro wrote: What I did once was create an announcement that got played to the receptionist announcing who the call was for based on the number that was called. This allowed the receptionist to know which greeting to recite. How did you do this so that the calling party does not hear the announcement? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Detecting Fax and bad CDRs
I used to own a company which made fax switches and yes your devise must "answer" the call to detect the fax CNG tone or wait till voicemail answers and try to detect the CNG tone while the voicemail greeting is being played. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Monday, May 02, 2005 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Detecting Fax and bad CDRs On Mon, 2005-05-02 at 21:21 -0500, Matthew Boehm wrote: > Here is the current scenario: > > Someone calls my DID. It comes in on PRI. I do "Answer()" then "Wait(1)" to > see if there is a fax tone. If there is, goto fax context and do the fax > thing. > > If no fax tone, then "Dial(SIP/myphone,30)". If I don't answer in the 30 > seconds, goto Voicemail. > > However! If the person were to hangup before the 30 seconds expired, the > channel would get hung up but the disposition in the CDR (along with the > billable seconds) would be inaccurate because I "Answer'd" the call > beforehand. > > Is there a way to change the disposition and the billable seconds to more > accurately reflect what happened? > > This really is only a big deal for inbound 800 calls; because if I answer > the call and it is a person and they hang up before voicemail picks up, I > have to charge the customer for those 10-15 seconds. Plus, I give customers > the ability to see their "missed" calls, which is based on a 0 billable > seconds duration in the CDR. There are two methods for detecting a fax (that I know of). 1) Allocate a specific DID for your faxes, so any call to that number will be a fax 2) Answer the call, listen for the fax tone, and if you receive it, then treat as a fax, else carry on. AFAIK, you can't listen to the audio prior to answering the call. (You can sometimes send audio prior to answering the call). So you can't tell if it is a fax or not. Personally, I presume you would need to bill your user for that 15 seconds, or else you will end up losing money. You answered the call, therefore the telco is charging you, if you don't charge your user, then you will lose. Of course, I could be wrong, but I doubt it :) I'd like to know how those 'home/soho' fax machines 'detect' an incoming fax and allow the inside phone to ring, and only pick up if it is a fax call. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
trixter http://www.0xdecafbad.com wrote: I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. Those were my exact thoughts. Had I received a warning, I might have been able to do something about it.. like set call forwarding to go to my cell phone until they worked out their kinks. Unprofessional in my view. A response to their email, even an auto responder noting an outage in some area would keep me from getting so hot under the collar. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. They do have enough information for most on their web site. The account portal seems put together enough. JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] signaling table of E100P Digium Cards
On Tue, 3 May 2005, James Lin wrote: > I am trying to configure an E100P channel bank card. > > The * will be connected to a PSTN switch with an E1 line. > > I am a bit confused with signaling table of E100P Digium Cards. > > The signaling table of E100P Digium cards is each 64K channel's ABCD > bits which indicate the state for transmit signaling. > > My problem now is where to find the CAS(Channel Associated Signalling) > signaling ABCD bit table ? (TxABCD, RxABCD) > > How can I modify the ABCD value ? For CAS on an E1 you need MFC/R2 support. See http://www.soft-switch.org/unicall/installing-mfcr2.html. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
On Mon, 2005-05-02 at 12:40 -0700, Sean Kennedy wrote: > Adam Goryachev wrote: > >On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: > >The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) > >phones can all do what he wants. ie, have multiple lines with blinking > >red lights when a call arrives on that line. > >The polycom ip600 and cisco 7960 both have 6 lines available. > Ok, this is the first I've heard about it. Will the lights show call > status? As in, if the call is put on hold on one of those other > extensions, it will flash? Or go green ( or another color ) when a call > is connected on another extension? > > Basically a mimic of the partner ACS systems? > > To my knowledge, there is no such thing. Am I wrong? >From what I have heard, the snom phone is capable of this. I have also heard some people say they got it working on polycom phones, but I haven't been able to get it working. Though maybe I haven't tried as hard as I need to... I have seen many requests from many people asking how to make this work, and am yet to see a decent 'howto'... Hopefully, someone with this working on a polycom will chip in with what they needed to config in the xml files, and the * extensions.conf/sip.con files Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting Fax and bad CDRs
On Mon, 2005-05-02 at 21:21 -0500, Matthew Boehm wrote: > Here is the current scenario: > > Someone calls my DID. It comes in on PRI. I do "Answer()" then "Wait(1)" to > see if there is a fax tone. If there is, goto fax context and do the fax > thing. > > If no fax tone, then "Dial(SIP/myphone,30)". If I don't answer in the 30 > seconds, goto Voicemail. > > However! If the person were to hangup before the 30 seconds expired, the > channel would get hung up but the disposition in the CDR (along with the > billable seconds) would be inaccurate because I "Answer'd" the call > beforehand. > > Is there a way to change the disposition and the billable seconds to more > accurately reflect what happened? > > This really is only a big deal for inbound 800 calls; because if I answer > the call and it is a person and they hang up before voicemail picks up, I > have to charge the customer for those 10-15 seconds. Plus, I give customers > the ability to see their "missed" calls, which is based on a 0 billable > seconds duration in the CDR. There are two methods for detecting a fax (that I know of). 1) Allocate a specific DID for your faxes, so any call to that number will be a fax 2) Answer the call, listen for the fax tone, and if you receive it, then treat as a fax, else carry on. AFAIK, you can't listen to the audio prior to answering the call. (You can sometimes send audio prior to answering the call). So you can't tell if it is a fax or not. Personally, I presume you would need to bill your user for that 15 seconds, or else you will end up losing money. You answered the call, therefore the telco is charging you, if you don't charge your user, then you will lose. Of course, I could be wrong, but I doubt it :) I'd like to know how those 'home/soho' fax machines 'detect' an incoming fax and allow the inside phone to ring, and only pick up if it is a fax call. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A good SIP receptionist phone
We at snom would love to have a good LED integration with Asterisk. The current state seems to be a good start, but can use some improvements. What would be the best way to push this? Maybe sit together for a few days and work on the integration (doing some dirty hacks). Who would be the right person to talk to? Olle? CS > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Sean Kennedy > Sent: Monday, May 02, 2005 10:46 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] A good SIP receptionist phone > > Adam Goryachev wrote: > > >On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: > > > > > > > >>2) There isn't anything like what you want. I know, I want > the same > >>thing. There is no phone out there that will do this with any > >>protocol that asterisk uses. This is the one major failing of > >>asterisk ( and voip in general. I smell an oportunity for a phone > >>manufacture ), and what keeps it out of a lot of places. > >> > >> > > > >It's alright, you can come out from under your rock now > > > >The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) > >phones can all do what he wants. ie, have multiple lines > with blinking > >red lights when a call arrives on that line. > > > >The polycom ip600 and cisco 7960 both have 6 lines available. > > > >Regards, > >Adam > > > Ok, this is the first I've heard about it. Will the lights > show call status? As in, if the call is put on hold on one > of those other extensions, it will flash? Or go green ( or > another color ) when a call is connected on another extension? > > Basically a mimic of the partner ACS systems? > > To my knowledge, there is no such thing. Am I wrong? > > Sean > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
On Mon, 2005-05-02 at 12:59 -0600, Jeff Pratt wrote: > Pedro wrote: > > What I did once was create an announcement that got played to the > > receptionist announcing who the call was for based on the number that > > was called. This allowed the receptionist to know which greeting to > > recite. > > > > Why not turn that around? Have the receptionist record a greeting for > each company, IE: > > Hello this is company X, how may I help you? > Welcome to The Y Corporation, how may I direct your call? > Z Corp! How can I help you? > > that then gets Play()ed to the customer when the receptionist picks up > the call. Saves wear and tear on the receptionists voice, so they'll > thank you, and it gives them a hint as to whicch company has been called. A friend of mine used to work in a bank call centre here, and he said they used this to ensure that every call was answered professionally/etc, however, when the greeting is quite long, often people will attempt to interrupt 'you', and hence you may end up double-talking ie, your recording of you + you actually saying something However, I'd just think those people are plain rude (though waiting on hold for a long time is annoying also...) Finally, could this even be accomplished within the asterisk dialplan? ie, play a message when the dial is answered? Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
On Tue, 2005-05-03 at 00:09 -0500, Tim Connolly wrote: > No.. but... > In their defense though, Cisco sold us a million dollars in routers > taunting they could handle the load. 6 months later we were trading them in > for Junipers because they were only able to handle the load as long as it > was low in packet per second count. Sometime they just don't have a way to > real world test this stuff without throwing it into the wind to see if it > flies. Yes, I can see that, I dont know if that is the case or if its configuration changes. Either way they should anounce that they are upgrading if that is what they are doing. Given the short duration of the outages and their frequency over time I would expect that is what they are doing and its not failing equipment. > I do have two big complaints about Broadvoice. I keep getting billed > for calls that they claim are free calls on their website. I even > implemented their suggested dial plan for Asterisk. Second complaint is that > I can only register with one proxy at a time. Why shouldn't I be able to > register with all three? > dont forget about proxy.nyc.broadvoice.com which is more hidden.. proxy.lab.broadvoice.com I think its testing but who knows, broadvoice is alledgly lowell ma based (about 1 hour outside boston) and lab is near there based on traceroute. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki Trouble?
You can try posting your questions to http://www.asterisktalk.com - there are a couple of "asteriskers" lurking around from time to time. Cheers On 5/3/05, Callum McGillivray <[EMAIL PROTECTED]> wrote: > Yep - I think it's down. > > Mojo Jojo wrote: > > > Anyone having trouble getting to the Wiki? > > > > http://www.voip-info.org/wiki-Asterisk > > > > > > -- > > Private Label Wholesale Internet Access! > > http://www.YourOwnISP.com > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ast 1.0.7, IP-500's with unmanaged switch...remote end missing bits of audio
I thought I would throw this out there and see if anyone has any ideas...I have the same problem at 2 locations. The complaint from the users is that calls "cut out", "kinda like when you have spotty cell coverage". Doesn't seem to matter whether the call is incoming or outgoing, although it might be true that my users hear the remote party cut out, while the remote party doesn't notice the same from my users... Location 1: - SDSL 1.5 Mpbs with static IP, Netopia 4652 SDSL router (enabled "Prioritize Delay Sensitive Data" to recognize tos=lowdelay per Netopia support) - Dell PowerEdge SC420 with TDM04B (currently only using one port. the single analog line is call forward on busy to my IAX provider) - Asterisk CVS-v1-0-02/22/05 using IAX to connect to my provider over the public internet - I have run pings for an extended period of time against my provider's server and get no packet loss. - In IAX.conf: tos=lowdelay, jitterbuffer=yes, also enabled "Prioritize Delay Sensitive Data" on the Netopia to support tos=lowdelay per Netopia support - Average ping time to my provider: 160 ms with no packet loss - 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw only - Small Business Server 2003 set up as DC for the network - Two network laser printers - 24 port unmanaged switch (all phones are home run back to a patch panel, patched from there into a switch port. The DSL modem, printers and server are patched into the switch in the same way) - 8 pc's running XP Pro, all plugged into the switch port on the back of the IP-500's Location 2: - Full rate data T1 - Dell PowerEdge SC1420 with no TDM hardware at all (this location connects SIP directly to the T1 providers BroadSoft switch and does not go over the public internet) - Asterisk 1.0.7 - using SIP to connect with my provider (not across public internet, not natted since the Cisco IAD does the SIP mangling for us) - Average ping time to the broadsoft switch: 42 ms - 8 Polycom IP-500's running SIP 1.4.1.0040 and bootrom 2.6.1 using ulaw only - Small Business Server 2003 set up as DC for the network - One network printer - 24 port unmanaged switch (all phones are home run back to a patch panel, patched from there into a switch port. The DSL modem, printers and server are patched into the switch in the same way) - 8 pc's running XP Pro, all plugged into the switch port on the back of the IP-500's As you can see, the only commonalities are Dell hardware (but not models), Asterisk (but not versions), IP-500's (including sip and bootrom version), SBS 2003, 24 port unmanaged switch, the fact that all the pc's are plugged into the switch ports on the phones. Same symptoms at both locations. I cannot determine any specific causes (ie it doesn't seem to be inbound vs. outbound, etc). I have checked all the pc's for viruses and worms, changed switch ports, etc...the only theory I have right now is that since the Polycoms give priority to outbound phone traffic vs a connected PC, that outbound voice is getting the QOS it needs. Versus inbound voice which gets no priority treatment once it hits either LAN since the switch can't do any QOS. Am I on the right track with this theory? Do I need to try a managed switch, giving priority to voice to make sure that both incoming and outgoing voice packets are preferred? On a side note, at what point (size - number of clients) is a managed switch recommended? required? If I'm off base with the QOS theory, what else should I be looking at? Marty ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
No.. but... In their defense though, Cisco sold us a million dollars in routers taunting they could handle the load. 6 months later we were trading them in for Junipers because they were only able to handle the load as long as it was low in packet per second count. Sometime they just don't have a way to real world test this stuff without throwing it into the wind to see if it flies. I do have two big complaints about Broadvoice. I keep getting billed for calls that they claim are free calls on their website. I even implemented their suggested dial plan for Asterisk. Second complaint is that I can only register with one proxy at a time. Why shouldn't I be able to register with all three? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, May 02, 2005 11:59 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? On Mon, 2005-05-02 at 21:39 -0700, Luki wrote: > has issues for 15 minutes. Stuff happens; I guess they are > implementing the required wire-tapping and/or 911 stuff... or > whatever. I think its fair to say they are doing something :) I also have few problems with them in general, just stuff happened recently, and while for some things, temporary outages arent that big of a deal, for many telephone stuff is more critical to their business or whatever. VoIP in general is new, and the legislation for US based (or those that operate there) companies is changing which is going to force them to make infrastructure changes. Depending on how it was originally set up it depends on how severe that is. I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
-BEGIN PGP SIGNED MESSAGE- On 09:50 PM 5/2/2005, Matthew Boehm wrote: > >Hold up. So you have Phone #1. And all 6 lines register with the username of >"phone1" ? > >And you have phone #2; and all 6 lines register with username of "phone2"? > >And the phone only registers once? Interesting..I'm gonna test this. Sounds >like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQncHJ+0LTNca2q41AQHf/QP7BB5ni6GOzc7JxvavF+ryg172gBtlIWku hmw5JkcinUBGKcRQ9paMXcZ+NRMokUFyljF+Yc1xLWPp4Gt1u/PCYmnU2tO/RIYg JAffPN5fVcA5zq5+uw/n0utwpUpo0VTzKPErcHonLJrr+ZF7MYxIiQ3NxHpQjAeR zOZw81xfyvU= =vHoj -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD Compatability
On Mon, 2005-05-02 at 21:55 -0700, snacktime wrote: > Here is the translation table from a dell poweredge 2650/3.06 mhz > running fbsd 5.4-stable. > > I'm running the zaptel ztdummy and * from 04/08/05 cvs head. Haven't > had any issues at all, but I'm not doing a lot of traffic either. > > Chris I am managing two systems, one debian linux on a low end athalon and the other fbsd 5.x on a p4 2.8GHz. The p4 is about 2x the speed of the athalon yet it has higher trnaslation times for all codecs, many are 1.5 times as long as the athalon which runs at about half the clock speed. fbsd was built from ports (1.0.7) the debian box was the debian sarge package (1.0.7). Both run fine, for the traffic I do, I just noticed that the translation times were higher on the faster box. Although I didnt build it I dont know if something odd was done. I also dont know if trnslation tables are computed realtime or not. If they are precomputed then something done when they were made could affect the times and thus they arent 'real'. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki Trouble?
Yep - I think it's down. Mojo Jojo wrote: Anyone having trouble getting to the Wiki? http://www.voip-info.org/wiki-Asterisk -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how stable is oh323 ?
how stable is oh323 ? is there any production implement ? ( which version 0.65 or 0.7?) could you share the experience ? Mario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
On Mon, 2005-05-02 at 21:39 -0700, Luki wrote: > has issues for 15 minutes. Stuff happens; I guess they are > implementing the required wire-tapping and/or 911 stuff... or > whatever. I think its fair to say they are doing something :) I also have few problems with them in general, just stuff happened recently, and while for some things, temporary outages arent that big of a deal, for many telephone stuff is more critical to their business or whatever. VoIP in general is new, and the legislation for US based (or those that operate there) companies is changing which is going to force them to make infrastructure changes. Depending on how it was originally set up it depends on how severe that is. I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wiki Trouble?
Anyone having trouble getting to the Wiki? http://www.voip-info.org/wiki-Asterisk -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD Compatability
On 5/2/05, trixter http://www.0xdecafbad.com <[EMAIL PROTECTED]> wrote: > On Mon, 2005-05-02 at 00:06 -0400, skamp wrote: > > asterisk runs great on BSD if you follow the sirections, and the card i > > believe does work > > That is debateable :) First which bsd? FreeBSD 4.x for example doesnt > compile becuase of the zaptel stuff, however 5.x does (from ports > anyway). In FBSD 5.x the translation times between codecs is higher > than a *slower* linux box (again fbsd 5.x ports vs 1.0.7 stable from the > debian sarge repository). The speed difference in cpu was roughly 2x > faster for fbsd (without anything running no load to speak of) and > translation times were 1.5 longer than the linux box. I was amazed at > that becuase I see no reason why the os would matter for that. Maybe > its just a display error from the CLI, where values were miscalculated > and stored and then when displayed incorrect. Here is the translation table from a dell poweredge 2650/3.06 mhz running fbsd 5.4-stable. g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 3 2 1 4 82312 ulaw - 3 - 1 3 2 1 4 82312 alaw - 3 1 - 3 2 1 4 82312 g726 - 4 3 3 - 3 2 5 92413 adpcm - 3 2 2 3 - 1 4 82312 slin - 2 1 1 2 1 - 3 72211 lpc10 - 3 2 2 3 2 1 - 82312 g729 - 3 2 2 3 2 1 4 -2312 speex - 3 2 2 3 2 1 4 8 -12 ilbc - 3 2 2 3 2 1 4 823 - I'm running the zaptel ztdummy and * from 04/08/05 cvs head. Haven't had any issues at all, but I'm not doing a lot of traffic either. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Hold up. So you have Phone #1. And all 6 lines register with the username of "phone1" ? And you have phone #2; and all 6 lines register with username of "phone2"? And the phone only registers once? Interesting..I'm gonna test this. Sounds like it'd be a solution to 1 of my many problems. -Matthew > From: Henry Devito <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Mon, 2 May 2005 22:50:24 -0500 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration > > Usernames the same. The Cisco phone recognizes the usernames are the same > and only registers once. > - Original Message - > From: "Matthew Boehm" <[EMAIL PROTECTED]> > To: "Asterisk Users" > Sent: Monday, May 02, 2005 8:36 PM > Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration > > >> You can't use the same extension on multiple line buttons but you can >>> Yes you can with the 7940's and 7960's. It works fine for a lot of >>> people >>> so calls will "roll" to the next button. I have this set up on several >>> customer sites. >> >>You can use the same "extension" but the usernames must be different. >> Right? >> >> -Matthew >> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Voicemail Access
Lenny, here's a sample:
[macro-voicemail]
exten => s,1,MailboxExists(${ARG1})
exten => s,2,Congestion()
exten => s,102,VoiceMail(su${ARG1})
exten => s,103,Hangup()
exten => a,1,VoiceMailMain()
exten => s,2,Hangup()
Call it like this:
exten => 1234,1,Dial(SIP/${EXTEN},30)
exten => 1234,2,Macro(voicemail,${EXTEN})
--Luki
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
> No, you can use the same username and secret for all 6 lines > > A 7940 or 7960 will just "do the right thing" > > That right thing being, roll over the new call to the second line if > the > first is busy, etc. There's no way. Asterisk doesn't support multiple logins from the same SIP username. You're telling me you have the same exact SIP.cnf file loaded into every one of your 7960's? You have 6 separate username/passwords and all 6 of those are registered on all of your Cisco 7960 phones? And that you can send a call to a specific extension on a specific phone? I'd have to see that to believe it because every developer on the -dev list will tell you that asterisk doesn't support simultaneous SIP logins. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
> I wonder how broadvoice is? LOL Quite frankly, they still are on my personal "recommended" list. Last week's downtime (those few hours when nothing worked, remember?) was the first outage this year for me. And yes, I'm using those accounts actively, each has easily 10-15 calls a day. I don't call today an outage since 10 out of 12 numbers worked for me, and those two only has issues for 15 minutes. Stuff happens; I guess they are implementing the required wire-tapping and/or 911 stuff... or whatever. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD Compatability
On Mon, 2005-05-02 at 00:06 -0400, skamp wrote: > asterisk runs great on BSD if you follow the sirections, and the card i > believe does work That is debateable :) First which bsd? FreeBSD 4.x for example doesnt compile becuase of the zaptel stuff, however 5.x does (from ports anyway). In FBSD 5.x the translation times between codecs is higher than a *slower* linux box (again fbsd 5.x ports vs 1.0.7 stable from the debian sarge repository). The speed difference in cpu was roughly 2x faster for fbsd (without anything running no load to speak of) and translation times were 1.5 longer than the linux box. I was amazed at that becuase I see no reason why the os would matter for that. Maybe its just a display error from the CLI, where values were miscalculated and stored and then when displayed incorrect. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: LiveVOIP
An update: I certainly cannot complain of lack of customer service or responsiveness. Yes, I'm a small customer for now, but considering that I've been getting great attention... at least more prompt and useful responses I'm used to getting from Broadvoice's support. At this point the issue remains unresolved, however, I'm told they are opening a Texas node soon, and since my server is in Texas that should be worth trying out for me. And to [EMAIL PROTECTED]: > switch to real provider That really isn't a constructive response... --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD Compatability
It's just that this statement from the wiki confused me: Asterisk is known to run on many OS platforms. However, Linux is the main platform for development and Digium hardware support. If you are running VoIP only, or if you are comfortable with using external media gateways to connect conventional telephone equipment, then you have more systems to choose from, like FreeBSD, Mac OS X and Solaris It sounds as if BSD-like OS are good to run asterisk without the digium boards. Thanks, - Daniel On May 2, 2005, at 12:06 AM, skamp wrote: asterisk runs great on BSD if you follow the sirections, and the card i believe does work On Tue, 2005-05-03 at 00:01 -0400, Daniel Salama wrote: Anyone know if Digium cards, especifically TE410P, are compatible with BSD (FreeBSD or NetBSD)? How does * run on BSD? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- skamp <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Taking asterisk out of the media path - SIP - howis it achieved
the following are necessary conditions for asterisk out of mediia path - - the codecs on both endpoints of the call should be the same. - the technology on both endpoints should be the same. - canreinvite=yes - no 't', 'T', L, conference, transfer flags in Dial command tulika From: David John Walsh <[EMAIL PROTECTED]> Reply-To: David John Walsh <[EMAIL PROTECTED]>,Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Taking asterisk out of the media path - SIP - howis it achieved Date: Mon, 2 May 2005 21:41:49 +0100 Hello How do you make asterisk stay out of the media stream? i.e once I set a call up between two parties, even if asterisk fell over the call would continue (in the same way a HLR on a mobile network works) I understand that many features will be lost if I do this, but all that I need seems to be supported by the end user hardware. incidentally I have tried canreinvite=yes, doesn't seem to work. I have also tried removing any flags in the dial() command Thank you for any information. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Bought a New Cellphone? http://adfarm.mediaplex.com/ad/ck/4686-26272-10936-265?ck=Register Sell your old one for a Great Price in eBay! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regarding asterisk-dev list
Hi, Sorry this is the wrong place to ask but I have subscribed to the dev list. I receive the posting, by my posts don't appear on the list. Am I missing something? Is there something more I need to do to be able to post on the dev list? Again sorry for this post but I thought I could get quick help. Thanks and regards, - Ashish ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940G
We have an interesting issue with relation to Confrence Calling on the 7940's. If a call is made to an internal number say "004" and then if we try and confrence a local number say "30184200" it dials it out via the international provider. However, just dialling "30184200" it dials it out via the correct provider. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Voicemail Access
Doesn't Voicemail already do this? I just tried with mine and pressing * while hearing the greeting dropped me into VoicemailMain for that box. Peter. On Mon, 2005-05-02 at 19:48 -0500, Lenwood S Sawyer III wrote: > All, > > Can a user dial in from the outside on a DID and then when they hear > there voicemail greeting press a key (maybe * key) and then be dumped > into the voicemail menu to check their messages? It seems that there is > a way using the "a" extension, but I can not find any detail on it? Any > help would be greatly appreciated. > > Thanks! > Lenny Sawyer > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD Compatability
asterisk runs great on BSD if you follow the sirections, and the card i believe does work On Tue, 2005-05-03 at 00:01 -0400, Daniel Salama wrote: > Anyone know if Digium cards, especifically TE410P, are compatible with > BSD (FreeBSD or NetBSD)? How does * run on BSD? > > Thanks, > Daniel > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- skamp <[EMAIL PROTECTED]> signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help getting zap trunk to work
Just to let you know, V 1.0 does fix the ZAP channel problem, although the trunk still does not show in FOP. Mike On Tue, 2005-05-03 at 07:57, Mike Price wrote: > [EMAIL PROTECTED] hove now released version 1.0. According to the changelog > this is supposed to fix the ZAP problem. I am trying it myself today, > for the same reason. I'll post my result. > > Mike > > On Tue, 2005-05-03 at 06:35, Listacc wrote: > > Hello All, > > > > I am currently using Asterisk @ Home 9.0 and I have the Digium Wildcard > > X100 / X100 I purchase off ebay, a old PIII and nic. I can get the trunk > > to show up in the flash operator panel thing or be active I dont know > > what it is but it doesn't show up once it is add and created and i even > > setup the out going plans etc. > > > > Regards, > > > > Otis Surratt Jr. > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BSD Compatability
Anyone know if Digium cards, especifically TE410P, are compatible with BSD (FreeBSD or NetBSD)? How does * run on BSD? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF in Voicemail
Is anyone aware of any fixes to DTMF in voicemail after CVS head 11/15/04. I have seen a few other posts about dtmf failing in voicemail and it seems in a least one other post the CVS date was around 11/04. We use snom phones with cvs 11/15/04 dtmfmode=rfc2833 If there are fixes an upgrade would be the way to go, but everything else works now so I do not want to move forward on a newer cvs an introduce other issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: LiveVOIP
well if it helps any i had a terrible time getting to sites in the US from the Philippines this morning. Though i could get to ASIA/EU sites fine On Mon, 2005-05-02 at 20:43 -0700, Chris Coulthurst wrote: > Did any Level 3 backbone go down about the same time? Maybe some BGP4 > routes got 'lifeboated'. > > Chris Coulthurst > [EMAIL PROTECTED] > > > |-Original Message- > |From: [EMAIL PROTECTED] [mailto:asterisk-users- > |[EMAIL PROTECTED] On Behalf Of David Josephson > |Sent: Monday, May 02, 2005 8:18 PM > |To: asterisk-users@lists.digium.com > |Subject: [Asterisk-Users] Re: LiveVOIP > | > |Luki writes about choppy audio with LiveVOIP. We have an almost > |identical situation except that we were switched from the San Diego > |gateway to the Van Nuys gateway. Some improvement but still not usable > |for real customers. I have an open trouble ticket with them and no > |progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming > |audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, > at > |best one dropout every 10 seconds, usually one short dropout every one > |to three seconds. The comments from their tech support and CTO were > that > |they were aware of the problem and it was "a capacity issue" that they > |were working on. There is a separate problem in that ringback tone (or > |any other audio sent without answer supervision being active, > |apparently) is not played to the PSTN side. This is not unique to > |LiveVOIP and has been discussed (with its workarounds) before. I don't > |mind their brusque attitude or the lack of user-level support, but we > |won't be able to use their service if they can't fix the dropouts. > There > |is a lot of clatter here on the list about them not being a "real > |provider" but a lot of this is sour grapes from people reselling more > |expensive service. We'll see ... they don't have to be 100% facilities > |based to provide good service, but they do have to fix this issue. > |___ > |Asterisk-Users mailing list > |Asterisk-Users@lists.digium.com > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- skamp <[EMAIL PROTECTED]> signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Usernames the same. The Cisco phone recognizes the usernames are the same and only registers once. - Original Message - From: "Matthew Boehm" <[EMAIL PROTECTED]> To: "Asterisk Users" Sent: Monday, May 02, 2005 8:36 PM Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will "roll" to the next button. I have this set up on several customer sites. You can use the same "extension" but the usernames must be different. Right? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: LiveVOIP
Did any Level 3 backbone go down about the same time? Maybe some BGP4 routes got 'lifeboated'. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of David Josephson |Sent: Monday, May 02, 2005 8:18 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Re: LiveVOIP | |Luki writes about choppy audio with LiveVOIP. We have an almost |identical situation except that we were switched from the San Diego |gateway to the Van Nuys gateway. Some improvement but still not usable |for real customers. I have an open trouble ticket with them and no |progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming |audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at |best one dropout every 10 seconds, usually one short dropout every one |to three seconds. The comments from their tech support and CTO were that |they were aware of the problem and it was "a capacity issue" that they |were working on. There is a separate problem in that ringback tone (or |any other audio sent without answer supervision being active, |apparently) is not played to the PSTN side. This is not unique to |LiveVOIP and has been discussed (with its workarounds) before. I don't |mind their brusque attitude or the lack of user-level support, but we |won't be able to use their service if they can't fix the dropouts. There |is a lot of clatter here on the list about them not being a "real |provider" but a lot of this is sour grapes from people reselling more |expensive service. We'll see ... they don't have to be 100% facilities |based to provide good service, but they do have to fix this issue. |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
I've only had teliax for about 2 weeks or so, and was impressed at first, but their complete LACK of customer support is truly sad. I think it's a one-man operation and he has a second job or something. Never a callback until a threatening email, and maybe a few hours in the morning is all Ive seen someone on the 'java chat' tech window. I'm already looking for a replacement. (I wonder how broadvoice is? LOL) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Mark Musone |Sent: Monday, May 02, 2005 8:12 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? | |Don't say that! i just moved from broadvoice to teliax!!! | | | |On 5/2/05, Andre Normandin <[EMAIL PROTECTED]> wrote: |> Mine seemed to have come back somewhere between 5:30 and 6... |> |> -Original Message- |> From: [EMAIL PROTECTED] |> [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis |> Sent: Monday, May 02, 2005 8:18 PM |> To: asterisk-users@lists.digium.com |> Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? |> |> yes |> |> from sometime after 4 to around 5:30 pm. |> |> jerry |> |> ___ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users |> |> ___ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users |> |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Boy.. with all this discussion about broadvoice lately, makes me appreciate my nufone account.. :-) On Mon, 2005-05-02 at 19:46, Andre Normandin wrote: > Mine seemed to have come back somewhere between 5:30 and 6... > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis > Sent: Monday, May 02, 2005 8:18 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? > > > yes > > from sometime after 4 to around 5:30 pm. > > jerry > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Whitten <[EMAIL PROTECTED]> kFuQ Productions signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: LiveVOIP
David, The fact of the matter is that LiveVoIP has no customer service. They don't care about small users or asterisk users. Other providers have higher prices but offer real customer service. Go look at teliax they care about customers service. On Mon, 2005-05-02 at 22:17, David Josephson wrote: > Luki writes about choppy audio with LiveVOIP. We have an almost > identical situation except that we were switched from the San Diego > gateway to the Van Nuys gateway. Some improvement but still not usable > for real customers. I have an open trouble ticket with them and no > progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming > audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at > best one dropout every 10 seconds, usually one short dropout every one > to three seconds. The comments from their tech support and CTO were that > they were aware of the problem and it was "a capacity issue" that they > were working on. There is a separate problem in that ringback tone (or > any other audio sent without answer supervision being active, > apparently) is not played to the PSTN side. This is not unique to > LiveVOIP and has been discussed (with its workarounds) before. I don't > mind their brusque attitude or the lack of user-level support, but we > won't be able to use their service if they can't fix the dropouts. There > is a lot of clatter here on the list about them not being a "real > provider" but a lot of this is sour grapes from people reselling more > expensive service. We'll see ... they don't have to be 100% facilities > based to provide good service, but they do have to fix this issue. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do you get rid of Spawn's
Hi everybody, How do I get rid of spawn's ?
example
-- Executing Dial("Zap/23-1", "sip/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Accepting call from 'xxx3672728' to 'xxx2769906' on channel 0/23,
span 1
-- SIP/xxx.xxx.xxx.xxx-0adc is ringing
-- SIP/xxx.xxx.xxx.xxx-0adc answered Zap/23-1
-- Channel 0/23, span 1 got hangup
== Spawn extension (from-pstn, xxx2769906, 1) exited non-zero on
'Zap/23-1'
-- Executing Dial("Zap/23-1", "sip/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
-- Got SIP response 404 "No user [EMAIL PROTECTED] at this
server" back from xxx.xxx.xxx.xxx
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Must be number dependent -- out of 12 numbers on one server, 2 regs failed (404 not found) for about 15 minutes around 4:30pm PST, but are fine since. Other numbers remained unaffected and work find incoming and outgoing... BTW, all numbers have the same area code and exchange. Interesting... --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: LiveVOIP
Luki writes about choppy audio with LiveVOIP. We have an almost identical situation except that we were switched from the San Diego gateway to the Van Nuys gateway. Some improvement but still not usable for real customers. I have an open trouble ticket with them and no progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at best one dropout every 10 seconds, usually one short dropout every one to three seconds. The comments from their tech support and CTO were that they were aware of the problem and it was "a capacity issue" that they were working on. There is a separate problem in that ringback tone (or any other audio sent without answer supervision being active, apparently) is not played to the PSTN side. This is not unique to LiveVOIP and has been discussed (with its workarounds) before. I don't mind their brusque attitude or the lack of user-level support, but we won't be able to use their service if they can't fix the dropouts. There is a lot of clatter here on the list about them not being a "real provider" but a lot of this is sour grapes from people reselling more expensive service. We'll see ... they don't have to be 100% facilities based to provide good service, but they do have to fix this issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Don't say that! i just moved from broadvoice to teliax!!! On 5/2/05, Andre Normandin <[EMAIL PROTECTED]> wrote: > Mine seemed to have come back somewhere between 5:30 and 6... > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis > Sent: Monday, May 02, 2005 8:18 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? > > yes > > from sometime after 4 to around 5:30 pm. > > jerry > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Mine seemed to have come back somewhere between 5:30 and 6... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Monday, May 02, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? yes from sometime after 4 to around 5:30 pm. jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] signaling table of E100P Digium Cards
Hi all I am trying to configure an E100P channel bank card. The * will be connected to a PSTN switch with an E1 line. I am a bit confused with signaling table of E100P Digium Cards. The signaling table of E100P Digium cards is each 64K channel's ABCD bits which indicate the state for transmit signaling. My problem now is where to find the CAS(Channel Associated Signalling) signaling ABCD bit table ? (TxABCD, RxABCD) How can I modify the ABCD value ? Thanks for your help. These are my system info: [zaptel.conf] # E100P card span=1,0,0,cas,hdb3,crc4 fxoks=1-12 unused=13-15 unused=17-31 # # Wildcard X100P card fxsks=32 defaultzone=us loadzone=us [zapata.conf] [channels] musiconhold=default signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=default channel => 1-12 language=en signalling=fxs_ks channel => 32 Output of zttool: - Alarms Span OK Digium Wildcard E100P E1/PRA Card 0 OK Wildcard X101P Board 1 Current Alarms: No alarms. Sync Source:Internally clocked IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 12/ 0 112333 1234567890123456789012345789012 TxA TxB TxC TxD RxA RxB RxC RxD - [root at GW /]# lsmod Module Size Used byNot tainted wcfxo 9376 0 (unused) wct1xxp13184 0 (unused) zaptel179168 0 [wcfxo wct1xxp] [root at GW /]# cat /proc/interrupts CPU0 0: 68392 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 453361 XT-PIC wcfxo 5: 1048 XT-PIC eth0 8: 1 XT-PIC rtc 9: 0 XT-PIC usb-ohci, usb-ohci, ehci-hcd 10: 415190 XT-PIC t1xxp 12: 32 XT-PIC PS/2 Mouse 14: 5257 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 [root at GW /]# cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3//CRC4 1 WCT1/0/1 FXOKS 2 WCT1/0/2 FXOKS ... 11 WCT1/0/11 FXOKS 12 WCT1/0/12 FXOKS -- James Lin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] processing power measurement?
Thank you very much Andrew, exactly what I was after. I think I'm wasting my time with more ram. It peaked at only 10% for a single call - I only ever get 2 to 3 calls at a time so I should be fine. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Latham > Sent: Monday, May 02, 2005 10:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] processing power measurement? > > # top > > On 5/2/05, Dean Collins <[EMAIL PROTECTED]> wrote: > > > > > > > > I'm about to add more ram to my Asterisk server. > > > > > > > > Is there some way to measure the 'processing power' of my asterisk > server > > easily? I'd like to understand how the additional ram affects my server. > > > > > > > > > > > > Cheers, > > > > Dean > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > -- > > Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) > WWW: http://lathama.com > Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] > If any of the above are down we have bigger problems than my email! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
BroadVoice is once again having issues .. There tech support is also reporting issues with no ETR so they don't even know what's wrong .. Inbound calls are hit and miss ... Outbound calls are DOA. BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Gutowski Sent: Monday, May 02, 2005 7:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? I haven't lost my registration, but calls from Verizon, Nextel, and Sprint cell phones to my BV number (which worked in the past) are now getting a "This call cannot be completed as dialed, please check the number and try again" message. Calls from landlines seem to work fine, and my Rogers phone which is currently roaming on Cingular connects fine. Normally I would think that it was a routing issue (like people had with certain exchanges and Cingular in the past), but between three carriers from phones that have used this number daily for the past 2 months or so? Anyone else have this happen? I called BV support and they said "they must be dialing the number wrong" and didn't want to do anything even after I explained how that is not the case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I haven't lost my registration, but calls from Verizon, Nextel, and Sprint cell phones to my BV number (which worked in the past) are now getting a "This call cannot be completed as dialed, please check the number and try again" message. Calls from landlines seem to work fine, and my Rogers phone which is currently roaming on Cingular connects fine. Normally I would think that it was a routing issue (like people had with certain exchanges and Cingular in the past), but between three carriers from phones that have used this number daily for the past 2 months or so? Anyone else have this happen? I called BV support and they said "they must be dialing the number wrong" and didn't want to do anything even after I explained how that is not the case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
-BEGIN PGP SIGNED MESSAGE- On 06:36 PM 5/2/2005, Matthew Boehm wrote: > > >>> You can't use the same extension on multiple line buttons but you can >> Yes you can with the 7940's and 7960's. It works fine for a lot of people >> so calls will "roll" to the next button. I have this set up on several >> customer sites. > >You can use the same "extension" but the usernames must be different. >Right? > No, you can use the same username and secret for all 6 lines A 7940 or 7960 will just "do the right thing" That right thing being, roll over the new call to the second line if the first is busy, etc. - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQnbg1O0LTNca2q41AQEXPwP8DIZZo3W2vEZjVzmFUHa4J6uOHI/WB1JH FD8STuiXMb0g+KmABe89DDtR521q4Yqr4D230GxDKqNm4WzFmd5n39MAiIVTrgam sbrAiCGU6qw+15jer2KnuD5HAWLuJuHgcKG5tdrhogdLXBPQ8wI/Ng12lyTLLv6K BKY7RM3G9o8= =pd7W -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting Fax and bad CDRs
Here is the current scenario: Someone calls my DID. It comes in on PRI. I do "Answer()" then "Wait(1)" to see if there is a fax tone. If there is, goto fax context and do the fax thing. If no fax tone, then "Dial(SIP/myphone,30)". If I don't answer in the 30 seconds, goto Voicemail. However! If the person were to hangup before the 30 seconds expired, the channel would get hung up but the disposition in the CDR (along with the billable seconds) would be inaccurate because I "Answer'd" the call beforehand. Is there a way to change the disposition and the billable seconds to more accurately reflect what happened? This really is only a big deal for inbound 800 calls; because if I answer the call and it is a person and they hang up before voicemail picks up, I have to charge the customer for those 10-15 seconds. Plus, I give customers the ability to see their "missed" calls, which is based on a 0 billable seconds duration in the CDR. Is there no better way to detect fax other than "answering" the call in the dialplan? Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 & Shorewall Install
Not sure about FC3 issues, but I use Shorewall on the border. My * server is a 1-1 NAT inside. Here's all I need in /etc/shorewall/rules: # for SIP, IAX2, IAX, RTP, MGCP ACCEPT net loc:192.168.1.5 udp 5060,4569,5036,1:2,2727 - - - - I probably don't need all that, since I'm not running some of it, but it works for me. If your firewall is your * machine, you probably will need something like "$FW" instead of "loc:192.168.1.5", and maybe "all" instead of "net". HTH, (De nada) - Dan Anonymous Account wrote: >Dear asterisk-users, > >Allow me to preface this newbie's question with a statement: > > 1. I searched the archives & the Wiki > 2. I Googled until I couldn't Google anymore > >My questions concern the installation of the latest/greatest Asterisk >on Fedora Core 3 with a Shorewall (Shoreline) Firewall installed. > >I haven't been able to find a step-by-step howto that is CURRENT that >addresses this particular configuration. Does anyone have a link they >could point me to? Please keep in mind the word "current" and by that >I mean something that takes into account that I am using a Kernel that >is 2.6+ and that Shorewall is version 2.2+ > >Mucho Gracias, amigos! > >031547 > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Probably a coincidence, but Teliax has been very choppy audio today. Most of my callers from 1 to 4 pm were Mr. Roboto. Please let it NOT be a trend... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jerry Geis |Sent: Monday, May 02, 2005 5:18 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? | |yes | |from sometime after 4 to around 5:30 pm. | |jerry | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] processing power measurement?
# top On 5/2/05, Dean Collins <[EMAIL PROTECTED]> wrote: > > > > I'm about to add more ram to my Asterisk server. > > > > Is there some way to measure the 'processing power' of my asterisk server > easily? I'd like to understand how the additional ram affects my server. > > > > > > Cheers, > > Dean > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Group
I'm looking at the SIP callgroup feature but I'm not sure if that's what I want. Maybe someone can point me in the right direction. I want to know if SIP has a similar feature as the Zap groups. I would like to be assign SIP entries into a group and then call the group or route calls to the group. It would behave the same way where it can choose the next available SIP entry from the group. Is this possible, as is? -- Richard Cook [EMAIL PROTECTED] T: 705-223-2000 ext 2010 <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CDR - Mysql
If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does "cdr mysql status" do? If it says "no such command" then you haven't loaded the cdr module. Did you do "make install" inside the asterisk-addons dir? Do you have autoload => yes in your modules.conf? -Matthew > From: Callum McGillivray <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Tue, 03 May 2005 12:03:17 +1000 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql > > Which Debug log ? Where is it located ? > > I can't see anything obvious that shows this info. > > Cheers, > > Callum > > (P.S. I'm not seeing a connection on the mySQL DB from the asterisk > machine, and I assumed that there should be one... what am I missing > here ? ) > > Matthew Boehm wrote: > >> What is in your debug log? It will show the exact SQL that is being >> executed. >> >> -Matthew >> >> >> >> >>> From: Callum McGillivray <[EMAIL PROTECTED]> >>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion >>> >>> Date: Tue, 03 May 2005 11:35:52 +1000 >>> To: >>> Subject: [Asterisk-Users] Asterisk CDR - Mysql >>> >>> Hi All, >>> >>> We have configured our Asterisk Server (CVS Head) to use mysql for >>> CDR's, following the guidelines located at >>> http://www.voip-info.org/wiki-Asterisk+cdr+mysql . >>> >>> When Asterisk starts up there are no errors, when we make a call there >>> are no errors, however I am not seeing records in the database. >>> >>> Any idea how what I should be looking for here? I'm a bit lost. >>> >>> Cheers, >>> >>> Callum >>> ___ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CDR - Mysql
Which Debug log ? Where is it located ? I can't see anything obvious that shows this info. Cheers, Callum (P.S. I'm not seeing a connection on the mySQL DB from the asterisk machine, and I assumed that there should be one... what am I missing here ? ) Matthew Boehm wrote: What is in your debug log? It will show the exact SQL that is being executed. -Matthew From: Callum McGillivray <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Tue, 03 May 2005 11:35:52 +1000 To: Subject: [Asterisk-Users] Asterisk CDR - Mysql Hi All, We have configured our Asterisk Server (CVS Head) to use mysql for CDR's, following the guidelines located at http://www.voip-info.org/wiki-Asterisk+cdr+mysql . When Asterisk starts up there are no errors, when we make a call there are no errors, however I am not seeing records in the database. Any idea how what I should be looking for here? I'm a bit lost. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CDR - Mysql
What is in your debug log? It will show the exact SQL that is being executed. -Matthew > From: Callum McGillivray <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Tue, 03 May 2005 11:35:52 +1000 > To: > Subject: [Asterisk-Users] Asterisk CDR - Mysql > > Hi All, > > We have configured our Asterisk Server (CVS Head) to use mysql for > CDR's, following the guidelines located at > http://www.voip-info.org/wiki-Asterisk+cdr+mysql . > > When Asterisk starts up there are no errors, when we make a call there > are no errors, however I am not seeing records in the database. > > Any idea how what I should be looking for here? I'm a bit lost. > > Cheers, > > Callum > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
>> You can't use the same extension on multiple line buttons but you can > Yes you can with the 7940's and 7960's. It works fine for a lot of people > so calls will "roll" to the next button. I have this set up on several > customer sites. You can use the same "extension" but the usernames must be different. Right? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and USRobotics Courier V.Everything
On May 2, 2005 08:53 pm, Eric Wieling aka ManxPower wrote: > Rumor has it that the chan_modem_* modules were written by a couple of > drunken college students over Spring Break and never touched again. You You weren't one of them, were you? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk CDR - Mysql
Hi All, We have configured our Asterisk Server (CVS Head) to use mysql for CDR's, following the guidelines located at http://www.voip-info.org/wiki-Asterisk+cdr+mysql . When Asterisk starts up there are no errors, when we make a call there are no errors, however I am not seeing records in the database. Any idea how what I should be looking for here? I'm a bit lost. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVOIP troubleshooting
switch to real provider On Mon, 2005-05-02 at 20:21, Luki wrote: > Hi everyone, > > I need some ideas to troubleshoot this issue: I recently got an 800 > numbers from LiveVOIP and it works but on most calls the caller gets > hears choppy audio (one drop out per 10 seconds or so). > > I know this isn't LiveVOIP's support forum but I'm sure some here use > their 800 service and I'm interested in their feedback and ideas. And > don't get me wrong, LiveVOIP's support has been quite good -- > cooperative, fast response, action taken as requested -- but I do not > want to try their patience. At this point I am not blaming them for > this issue either. > > Here's the summary: > > * I'm connected via IAX2 to > * The server is in a datacenter with plenty of bandwidth. > * Using ulaw with "standard" 20 ms frames. > * I hear the caller perfectly fine, caller hears choppy audio. > * tcpdump shows incoming and outgoing packets right on time, > every 20 ms in each direction. > * I'm not using trunking for now. > * Pings to LiveVOIP are about 35 ms. > * iax2 show channels shows 1 ms jitter, 42 ms lag. > * Drop outs occur on IVR (or audio generated on the server itself) or > during normal conversation with a SIP client (ATA or phone) connected > to the server remotely. Connection between server and phones is well > tested and working fine. > > I have asked LiveVOIP to switch me from their Vancouver node to their > New York node, which reduced ping times from 50 ms to 35 ms. Less > chops but still not perfect. > > Note that the same server is already connected to several Broadvoice > accounts, which work flawlessly. > > Anyway, if anyone has some ideas of what I can try, please let me > know. I do not want to keep trying all their nodes to find one that > works for me. I do not necessarily want to use a different codec > either since I have the bandwidth and I may be receiving faxes, so I > need ulaw. > > Thanks and sorry for the long-ish post. > --Luki > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LiveVOIP troubleshooting
Hi everyone, I need some ideas to troubleshoot this issue: I recently got an 800 numbers from LiveVOIP and it works but on most calls the caller gets hears choppy audio (one drop out per 10 seconds or so). I know this isn't LiveVOIP's support forum but I'm sure some here use their 800 service and I'm interested in their feedback and ideas. And don't get me wrong, LiveVOIP's support has been quite good -- cooperative, fast response, action taken as requested -- but I do not want to try their patience. At this point I am not blaming them for this issue either. Here's the summary: * I'm connected via IAX2 to * The server is in a datacenter with plenty of bandwidth. * Using ulaw with "standard" 20 ms frames. * I hear the caller perfectly fine, caller hears choppy audio. * tcpdump shows incoming and outgoing packets right on time, every 20 ms in each direction. * I'm not using trunking for now. * Pings to LiveVOIP are about 35 ms. * iax2 show channels shows 1 ms jitter, 42 ms lag. * Drop outs occur on IVR (or audio generated on the server itself) or during normal conversation with a SIP client (ATA or phone) connected to the server remotely. Connection between server and phones is well tested and working fine. I have asked LiveVOIP to switch me from their Vancouver node to their New York node, which reduced ping times from 50 ms to 35 ms. Less chops but still not perfect. Note that the same server is already connected to several Broadvoice accounts, which work flawlessly. Anyway, if anyone has some ideas of what I can try, please let me know. I do not want to keep trying all their nodes to find one that works for me. I do not necessarily want to use a different codec either since I have the bandwidth and I may be receiving faxes, so I need ulaw. Thanks and sorry for the long-ish post. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and USRobotics Courier V.Everything
I could try to study the sources, however I did not find them. It knows to say me as I can find them? thanks Guillerme Em Seg, 2005-05-02 às 19:53 -0500, Eric Wieling aka ManxPower escreveu: > Guilherme Baião wrote: > > hi for all, would like to know if asterisk supports the modem USRobotics > > Courier V.Everything. Looking for very I found one link that it says on > > a called module chan_modem_usr2976.so that the principle would function, > > but lowering the sources of asterisk I did not find this module, > > somebody can help me? > > Rumor has it that the chan_modem_* modules were written by a couple of > drunken college students over Spring Break and never touched again. You > are welcome to update them to make them work. However, you'll have to > learn C as well as coding channel modules. > > Why have these not been removed yet? > > --Eric > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
It is only registering with asterisk once. ASterisk doesn't know that it is a second button on the phone, the sip software on the phone makes the call roll to the next free button. - Original Message - From: "Matthew Boehm" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, May 02, 2005 5:08 PM Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
- Original Message - From: "Scott Henderson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, May 02, 2005 4:01 PM Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will "roll" to the next button. I have this set up on several customer sites. use different extensions on different line buttons. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as VM for Nortel System
Wont work on an option unless you are using some sort of voice bridge or T1 card. There is no disconnect supervision on a Nortel option PBX on analog ports. Until recently the only indication the call is over is silence, recent releases of option software for the 61 and 11 produce dialtone when the call ends. - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: Sent: Monday, May 02, 2005 12:39 PM Subject: Re: [Asterisk-Users] Asterisk as VM for Nortel System On May 2, 2005 01:01 pm, Henry Devito wrote: What type of Nortel system? Is it an option or a norstar? *should* work with either, but I've only got "screw around" access with a MICS. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and USRobotics Courier V.Everything
Guilherme Baião wrote: hi for all, would like to know if asterisk supports the modem USRobotics Courier V.Everything. Looking for very I found one link that it says on a called module chan_modem_usr2976.so that the principle would function, but lowering the sources of asterisk I did not find this module, somebody can help me? Rumor has it that the chan_modem_* modules were written by a couple of drunken college students over Spring Break and never touched again. You are welcome to update them to make them work. However, you'll have to learn C as well as coding channel modules. Why have these not been removed yet? --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Matthew Boehm wrote: Well that proves my point; If you have 6 7960's and want 6 lines on each phone, you will need 36 username entries in your sip.conf. ouch.. That's just the way it is. Some people have 1,000 or more entries in sip.conf. It's not like they change all that often. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External Voicemail Access
All, Can a user dial in from the outside on a DID and then when they hear there voicemail greeting press a key (maybe * key) and then be dumped into the voicemail menu to check their messages? It seems that there is a way using the "a" extension, but I can not find any detail on it? Any help would be greatly appreciated. Thanks! Lenny Sawyer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone else having Broadvoice issues today?
yes from sometime after 4 to around 5:30 pm. jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Its like they are rebooting their sip proxies, my sip connection has been up/down all day, when it starts to come back it returns a 404 and then finally registers. I think this is related to the upgrades they have been doing for about a week now, which I also believe are related to the FCC ruling that VoIP providers in the US who connect to the PSTN provide CALEA (wiretap) support or be fined. On Mon, 2005-05-02 at 16:00 -0700, JD Austin wrote: > > Andre Normandin wrote: > > >Hello, > > > >About 4PM EDT I noticed that my broadvoice service cannot register.. > > > >Anyone else having problems with their broadvoice service? > > > >FYI: I connect to the 147.135.20.128 (nyc) proxy... > > > >Thanks, > > - Andre > I'm down too. > > BROADVOICE do you watch this list? > This is twice in seven days that you've had an outage. > > JD > -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Event
Is there a way to configure asterisk to execute an AGI script upon the transferring of a call to an extension from the Queue? For example, once the call is put in the queue and the extension becomes available, the Queue app will send the call to that extension. Is there a way for me to manually execute a command that will give me the extension it was transferred to? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help getting zap trunk to work
ok sir thanks Mike Price wrote: [EMAIL PROTECTED] hove now released version 1.0. According to the changelog this is supposed to fix the ZAP problem. I am trying it myself today, for the same reason. I'll post my result. Mike On Tue, 2005-05-03 at 06:35, Listacc wrote: Hello All, I am currently using Asterisk @ Home 9.0 and I have the Digium Wildcard X100 / X100 I purchase off ebay, a old PIII and nic. I can get the trunk to show up in the flash operator panel thing or be active I dont know what it is but it doesn't show up once it is add and created and i even setup the out going plans etc. Regards, Otis Surratt Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and USRobotics Courier V.Everything
hi for all, would like to know if asterisk supports the modem USRobotics Courier V.Everything. Looking for very I found one link that it says on a called module chan_modem_usr2976.so that the principle would function, but lowering the sources of asterisk I did not find this module, somebody can help me? thanks for all Guilherme ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] automated availabilty testing
On May 2, 2005 07:20 pm, JD Austin wrote: > Has anyone figured out how to know if your voip connection goes down > without actually dialing in and out manually? > During all of the broadvoice outages I've had as far as asterisk was > concerned, I was registered. > It was only when I tried dialing in or out that it became obvious that > something was up. It ain't hard; periodically generate a .call file to make a test call and see if it works. The call doesn't have to be to a human; any old DID that you can do some testing with will work just fine. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Well that proves my point; If you have 6 7960's and want 6 lines on each phone, you will need 36 username entries in your sip.conf. ouch.. -Matthew Scott Henderson wrote: >> You can configure multiple button to use multiple extensions. You do >> this in the phones configure file, here is a quick example. >> >> In this example the phone is the receptionist for two companies and >> it >> also has an individual line appearance for the the receptionist DID >> line. Keep in mind that if you configure the voice mail box in the >> sip.conf fill then you will see a voice mail indicator for each line >> as well. >> >> You will need to create entries in the sip.conf file for each >> extension >> as well as dial plan entires in extensions.conf but then all should >> be well >> >> # SIP Configuration Generic File >> >> # Line 1 >> line1_name: CRV_Reception >> line1_authname: "crv_reception" >> line1_password: "crv_reception" >> >> # Line 2 >> line2_name: "KP_Reception" >> line2_authname: "kp_reception" >> line2_password: "kp_reception" >> >> # Line 3 >> line3_name: "Colleen" >> line3_authname: "colleen" >> line3_password: "colleen" >> >> # Line 4 >> line4_name: "Line 4" >> line4_authname: "UNPROVISIONED" >> line4_password: "UNPROVISIONED" >> >> # Line 5 >> line5_name: "Line 5" >> line5_authname: "UNPROVISIONED" >> line5_password: "UNPROVISIONED" >> >> # Line 6 >> line6_name: "Line 6 >> line6_authname: "UNPROVISIONED" >> line6_password: "UNPROVISIONED" >> >> Scott Henderson >> >> Finite Technologies Incorporated >> 3763 Image Drive, Anchorage, Alaska 99504 >> Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 >> http://www.finite-tech.com >> http://www.chillywall.com >> http://www.virtuale.cc >> http://www.mphage.com >> Current Local Time: >> http://www.worldtimeserver.com/time.asp?locationid=US-AK >> >> >> >> >> Matthew Boehm wrote: >> >>> Tony Hoyle wrote: >>> >>> Scott Henderson wrote: > You can't use the same extension on multiple line buttons but you > can use different extensions on different line buttons. > > > Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony >>> >>> Please explain in more detail Tony. I've got tons of 7960's and we >>> only use the first button because asterisk doesn't support multiple >>> SIp registrations. >>> >>> -Matthew >>> >>> ___ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> > > > >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf dial plan
Here isa quick example, this rings the reception only for 10s and then
rings reception and another phone for 15s and then finally the voicemail.
; Reception
exten => 8900,1,Dial(SIP/crv_reception,10,Ttr); SIP - Reception
exten => 8900,2,Dial(SIP/crv_reception&SIP/gayle,15,Ttr)
exten => 8900,3,Voicemail(u8900)
exten => 8900,4,Voicemail(b8900)
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
Georg P. Israel wrote:
Dear Asterisk users,
I was wondering if anybody can tell me how to define a dial scheeme such
that an incomming all first rings for e.g. 20 seconds on one set of
phones and then after this time extends it's range onto a bigger set of
phones.
Basically, this is easy,
I can do this in the "extensions.con" with
[ISDN-in]
exten=> 6201030,1,setcallerid("${CALLERID}" <${CALLERID}>|a)
exten=> 6201030,2,dial,${UserGroup1}|20|t
exten=> 6201030,3,dial,${UserGroup1&UserGroup2}|60|t
exten=> 6201030,4,Voicemail2(u6201030)
exten=> 6201030,5,hangup
exten=> 6201030,302,Voicemail2(b6201030)
But here is on major problem,
in step 2, after 20 seconds, the call on the phones in Group1 will be
terminated and then restarted in the bigger group (Group1&Group2).
The problem with this is, during the transition is a time gap of a view
seconds on the phones from Group1. That means, if I lift up the head set
during this gape, then I can loos the calls on those phones.
Hence, I was wondering if I can set the dial proceadure such, that I
have the calls for 80 seconds on the phone Group1, and after 20 seconds
additionally on the phone Group2 without any interruption of the ringing
on the other phones.
Best regards
Georg P. Israel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] automated availabilty testing
Has anyone figured out how to know if your voip connection goes down without actually dialing in and out manually? During all of the broadvoice outages I've had as far as asterisk was concerned, I was registered. It was only when I tried dialing in or out that it became obvious that something was up. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mp3 problems
Hi, I recompiled Asterisk 1.0.3 on a machine which I upgraded the kernel. I also recompiled zaptel and libpri. After doing this, I am realizing that I'm having some problems playing mp3 files. However, and very strangely, music on hold is working playing mp3 files. I have an AGI script that was working just fine. You would select a recording ID and it would go out and fetch it and then play the file. Now, it's doing everything as it should, but it's not playing the actual media. I turned on agi debug on and here is the relevant portion of what I saw: AGI Rx << EXEC MP3Player /var/spool/asterisk/monitor/archive/4082-20050426-143915 -- AGI Script Executing Application: (MP3Player) Options: (/var/spool/asterisk/monitor/archive/4082-20050426-143915) May 2 19:15:46 NOTICE[2110]: app_mp3.c:91 timed_read: Poll timed out/errored out with 0 AGI Tx >> 200 result=0 Any clues? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD My connection is back up. Maybe they DO read this list :) -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
You can configure multiple button to use multiple extensions. You do this in the phones configure file, here is a quick example. In this example the phone is the receptionist for two companies and it also has an individual line appearance for the the receptionist DID line. Keep in mind that if you configure the voice mail box in the sip.conf fill then you will see a voice mail indicator for each line as well. You will need to create entries in the sip.conf file for each extension as well as dial plan entires in extensions.conf but then all should be well # SIP Configuration Generic File # Line 1 line1_name: CRV_Reception line1_authname: "crv_reception" line1_password: "crv_reception" # Line 2 line2_name: "KP_Reception" line2_authname: "kp_reception" line2_password: "kp_reception" # Line 3 line3_name: "Colleen" line3_authname: "colleen" line3_password: "colleen" # Line 4 line4_name: "Line 4" line4_authname: "UNPROVISIONED" line4_password: "UNPROVISIONED" # Line 5 line5_name: "Line 5" line5_authname: "UNPROVISIONED" line5_password: "UNPROVISIONED" # Line 6 line6_name: "Line 6 line6_authname: "UNPROVISIONED" line6_password: "UNPROVISIONED" Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Matthew Boehm wrote: Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Andre Normandin wrote: Hello, About 4PM EDT I noticed that my broadvoice service cannot register.. Anyone else having problems with their broadvoice service? FYI: I connect to the 147.135.20.128 (nyc) proxy... Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * and Sipgate (UK)
On Friday 29 April 2005 23:20, Paul Tyreman wrote: > What are you using instead of SIPGATE in the UK ? > > I also have this problem with DTMF tones not being passed to Asterisk from > a PSTN line and my e-mails are being ignored too ! Both are often reported problems with Sipgate. I have the DTMF problem with them. They took a couple of weeks to reply to an email about something else. Now I use voiptalk.org Gary -- Gary Stimson Zedcore Systems UK t/f: 0114 238 1828 USA/Canada toll-free t/f: 1-877-ZEDCORE Global t/f: +44 114 238 1828 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please find me a IAX provider
Seems they have a problem today.. Im trying to dial 011525552339281 but says noone could answer when in fact I know there is somebody there... Also, trying to dial 1800 numbers in the us with same luck.. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Roman Zhovtulya |Sent: Lunes, 02 de Mayo de 2005 01:23 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Please find me a IAX provider | |www.voipjet.com is much cheaper, by the way (but they charge |per-minute) | | | | |> -Original Message- |> From: [EMAIL PROTECTED] |> [mailto:[EMAIL PROTECTED] On Behalf |Of Kanuri, |> Seshu (Company IT) |> Sent: Montag, 2. Mai 2005 19:24 |> To: Kumara Jayaweera; Asterisk Users Mailing List - Non-Commercial |> Discussion |> Subject: RE: [Asterisk-Users] Please find me a IAX provider |> |> |> In all probabilty you will be able to make just one |simulatneous call |> with that bandwidth, where you need two channels of 64 Kbps each in |> the two directions, using Ulaw ( assuming both users are |blabbering at |> the same time). |> |> You don't need any IAX service providers. You just need a |$10 Account |> with any SIP based VOIP Calling Card Company. |> |> I suggest that you sign up as a user for sip account at |> http://www.terracall.com This is simple and best. |> |> Seshu |> |> -Original Message- |> From: [EMAIL PROTECTED] |> [mailto:[EMAIL PROTECTED] On Behalf Of Kumara |> Jayaweera |> Sent: Monday, May 02, 2005 11:41 AM |> To: asterisk-users@lists.digium.com |> Subject: [Asterisk-Users] Please find me a IAX provider |> |> Hi all, |> I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with |> softphones. I don't need incoming calls (no need DIDs). |Could someone |> tell me who is the best IAX service provider for me? |> I want unlimited monthly basis or yearly basis service. my DSL is |> 128kbps. Thank You Kumara |> |> ___ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |>http://lists.digium.com/mailman/listinfo/asterisk-users |> |> |> NOTICE: If received in error, please destroy and notify sender. |> Sender does not waive confidentiality or privilege, and use is |> prohibited. |> |> ___ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |>http://lists.digium.com/mailman/listinfo/asterisk-users |> | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Joe and Charlie,
YES, that fixed the problem. I did move the whole network to G729 but
it was never a codec problem.
I'm not running CVS, it's 1.0.3 at the moment.
Thanks
Scott H
Joe Baptista wrote:
On May 2, 2005 10:31 am, Charlie Watts wrote:
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests,
but it is doing it now. I will try to track down the specific change
tonight ...
My solution for now is to Answer() the call before dialing out. I changed
all of my outbound dialing rules from:
Same problem encountered here. My solution is to answer and play a sec of
silence before the dial proceeds - if i don't answer both parties are
connected but can't hear each other.
joe
[trunklocal]
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten => _9NXX,1,Answer
exten => _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
I’m running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1’s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing. During
the time that the caller should hear ringing the * console produces pages
of errors.
…..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw Apr 30 08:41:03 NOTICE[2813]:
channel.c:1314 ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw …..
I have tested this with the phones behind a PIX firewall with NAT, behind a
PIX firewall without NAT, and without a firewall at all. Nat is not the
problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the
* server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten =>
--6081,1,Dial(SIP/--6081&SIP/--6091,20)
=
-- Executing Goto("SIP/---..241.35-40400490", "TPN|--6081|1") in
new stack -- Goto (TPN,--6081,1)
-- Executing Dial("SIP/---.---.241.35-40400490",
"SIP/--6081&SIP/--6091|20") in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 "Moved Temporarily" back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268) -- Executing
Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490 -- SIP/---.---.241.35-f522 answered
Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered
SIP/---.---.---.35-40400490 == Spawn extension (TPN, --6081, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2' -- Attempting native
bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==
Now here is the console output with a single phone defined in the
extensions.conf (exten => --6081,1,Dial(SIP/--6091,20)
*
Asterisk-A*CLI>
-- Executing Goto("SIP/---.---.241.35-40418730", "Charity|--3263|1") in
new stack -- Goto (Charity,---263,1)
-- Executing Dial("SIP/---.---.241.35-40418730", "SIP/--3263|18") in
new stack -- Called --3263
-- Got SIP response 302 "Moved Temporarily" back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/--3263-f670) -- Executing
Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of forma
RE: [Asterisk-Users] Debuging SIP
Ian. I have a nat router in front of asterisk and Im forward 5060 and 1-2 to the asterisk box. On the other isde there is also a nat and they have budgetones.. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Ian Pattison |Sent: Lunes, 02 de Mayo de 2005 03:03 p.m. |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] Debuging SIP | |Are you using IPTables? Here's my IPTables config for SIP and |RTP... you need to open UDP ports 5060 and 1-2 | |$IPTABLES -A FORWARD -i $EXTIF -o $INTIF -p udp -m udp --sport |5060 --dport 5060 -j ACCEPT $IPTABLES -A FORWARD -i $EXTIF -o |$INTIF -p udp -m udp --sport 1:2 --dport 1:2 |-j ACCEPT $IPTABLES -A FORWARD -i $INTIF -o $EXTIF -p udp -m |udp --sport 5060 --dport 5060 -j ACCEPT $IPTABLES -A FORWARD |-i $INTIF -o $EXTIF -p udp -m udp --sport 1:2 --dport |1:2 -j ACCEPT $IPTABLES -t nat -A PREROUTING -i $EXTIF |-p udp -m udp -s EXTERNAL_SIP_SOURCE --dport 5060 -j DNAT |--to-destination ASTERISK_SERVER $IPTABLES -t nat -A |PREROUTING -i $EXTIF -p udp -m udp -s EXTERNAL_SIP_SOURCE |--dport 1:2 -j DNAT --to-destination ASTERISK_SERVER | |Hope this helps, | |Ian | |Ian Pattison, Senior Analyst |Technology Associates Inc. |Tel: 905-459-2100 ext. 204 |Mobile: 416-568-6548 |E-mail: [EMAIL PROTECTED] |WWW: http://www.technologyassociates.ca | |>>> [EMAIL PROTECTED] 02/05/2005 14:44 >>> |Guys. | |Im having NAT problems. Any good tips on how to debug remote |SIPS, how to see which ports are been sent and received, etc? | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Debuging SIP
What are you guys doing when you have sip phones behind nat on a remote end and then nat and asterisk on another? Sip hardphone - nat - internet - nat -asterisk What setup are you using? Examples sip.conf and hardphone configs? Firewall settings? This is driving me crazy! I have qualify and nat?yes on sip.conf and when the hardphone calls the call comes thru but neither the caller or myself can heard each other... My nat side is forward port 5060 and 1-2 to asterisk... Any ideas? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Schulte |Sent: Lunes, 02 de Mayo de 2005 02:21 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Debuging SIP | |sip debug |sip debug peer username |sip debug peer ip_address | |-Original Message- |From: Anton Krall [mailto:[EMAIL PROTECTED] |Sent: Monday, May 02, 2005 1:44 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Debuging SIP | | |Guys. | |Im having NAT problems. Any good tips on how to debug remote SIPS, how |to see which ports are been sent and received, etc? | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone else having Broadvoice issues today?
Hello, About 4PM EDT I noticed that my broadvoice service cannot register.. Anyone else having problems with their broadvoice service? FYI: I connect to the 147.135.20.128 (nyc) proxy... Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing calls, X100P
I am in Turkey. I imagine this is due to incorrect zone information, but I can't seem to be able to find the correct values for Turkey. I tried guessing them with no luck. DTMF tones in Turkey are the same as the standards everywhere. The other signalling tones are different (such as dial tone), but irrelevant in your case. Iain's suggestion may be what you need. Soner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] processing power measurement?
I’m about to add more ram to my Asterisk server. Is there some way to measure the ‘processing power’ of my asterisk server easily? I’d like to understand how the additional ram affects my server. Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Tony Hoyle wrote: > Scott Henderson wrote: >> You can't use the same extension on multiple line buttons but you can >> use different extensions on different line buttons. >> > Actually you can, and the 7960 does the 'right thing'.. surprised me > too. > > Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Your message to Asterisk-Users awaits moderator approval
Is anyone actually doing this? -Matthew [EMAIL PROTECTED] wrote: > Your mail to 'Asterisk-Users' with the subject > > bad CLI colors > > Is being held until the list moderator can review it for approval. > > The reason it is being held: > > Message body is too big: 136701 bytes with a limit of 40 KB > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. If you would like to cancel > this posting, please visit the following URL: > > > http://lists.digium.com/mailman/confirm/asterisk-users/c848474a9fcc1378ec80261492565225d2bd44bc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Joseph wrote: > On Mon, 2005-05-02 at 21:51 +0200, Kristof Hardy wrote: >> Patrick M. Gray, Jr. wrote: >>> In google'ing around a bit, it seems I should be able to assign the >>> same extension to several of the SIP lines on the 7960, and >>> asterisk should > > Works fine for us. > > Do you get errors? If you have 6 of these phones, then that means you have 36 SIP username/password combinations right? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules
Anyone? dmesg is saying each module and card is detected and Digium tech support isn't helpful at all with this. Robert Webb wrote: On Wed, 27 Apr 2005 11:24:24 -0600 Andrew Elchuk <[EMAIL PROTECTED]> wrote: Hi, I have two of the above installed into a server running Asterisk on Debian Linux. Currently, only two phone lines are connected to the system. I had both phone lines plugged into the one card, and it worked fine for dialing out on them, but when receiving incoming calls, only the line plugged into port 1 would answer. I then tried plugging the other line into ports 3 and 4 on the first wildcard and they were no go, and then tried port 1 on the other card and it worked fine for dialing out and answering?? I checked and the cards are not sharing an IRQ with anything else, and in wcfxs.c, #define AUDIO_RINGCHECK 1 is already commented out, as other forums mentioned it could be a problem. What gives here? zaptel.conf fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=en context=main-menu signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 musiconhold=default callerid=asreceived signalling=fxs_ks channel => 1-8 Thanks. How about posting the appropriate lines from dmesg to make sure that all the channels were recognized by the zaptel driver. Then we can go from there. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 & Shorewall Install
El lun, 02-05-2005 a las 16:14, Anonymous Account escribió: > My questions concern the installation of the latest/greatest Asterisk > on Fedora Core 3 with a Shorewall (Shoreline) Firewall installed. For me * works fine, with FC3 and iptables (Shorewall is an IPtables too) > I haven't been able to find a step-by-step howto that is CURRENT that > addresses this particular configuration. The config is similar for all distros http://www.voip-info.org/wiki-Asterisk+Linux+Fedora > Does anyone have a link they > could point me to? Please keep in mind the word "current" and by that > I mean something that takes into account that I am using a Kernel that > is 2.6+ and that Shorewall is version 2.2+ You only need to open the ports, but is better to use iptables directly ;) http://www.voip-info.org/wiki-Asterisk+firewall+rules > Mucho Gracias, amigos! De nada, Saludos desde Peru, -- Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 3 & Shorewall Install
Dear asterisk-users, Allow me to preface this newbie's question with a statement: 1. I searched the archives & the Wiki 2. I Googled until I couldn't Google anymore My questions concern the installation of the latest/greatest Asterisk on Fedora Core 3 with a Shorewall (Shoreline) Firewall installed. I haven't been able to find a step-by-step howto that is CURRENT that addresses this particular configuration. Does anyone have a link they could point me to? Please keep in mind the word "current" and by that I mean something that takes into account that I am using a Kernel that is 2.6+ and that Shorewall is version 2.2+ Mucho Gracias, amigos! 031547 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
