RE: [Asterisk-Users] Music on hold & Read error on sound device
Your soundcard doesn't have much to do with MOH. In fact, you can run MOH entirely without a soundcard. Therefore, OSS vs. ALSA won't affect your MOH performance. You might want to make sure that your musiconhold.conf file is correct, and that you really do have MPG123 and NOT MPG321. res_musiconhold.c looks for mpg123 in /usr/bin, so make sure it's in that location. If everything is running well, you should see two mpg123 processes running per "class" when you start Asterisk. Have you tried using the MP3Player to make sure that mpg123 is working? -wade > You didn't mention the distro you are using. I'm wondering if you are > using one of the distros that leans towards the alsa drivers. If so, > then chan_oss would have problems. > > On Sun, 2003-07-20 at 19:16, Stuart Hirst wrote: > > I have installed mpg123 which seems to be working fine but when I > > start *, I get the following error message at the CLI prompt when I > > start *: > > > > WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error > > on sound device: Resource temporarily unavailable > > > > I have checked that the sound card works by loading X11 and running > > sound tests which is fine. I have used "lsof /dev/dsp" to see if > > another application or server is controlling the sound device and > > without * running, nothing is reported. With * running lsof reports > > that * has the device. Voicemail works fine. > > > > When I put a call on hold the CLI shows moh starting but nothing is > > played. No errors are reported whilst starting moh. > > > > I have been trying lots of different things for hours now without > > success. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Best" VoIP provider for Asterisk?
> "Dan" == Dan <[EMAIL PROTECTED]> writes: >> Agreed. Jeremy McNamara of Nufone.net is the top dog in Asterisk >> VOIP and long distance. Dan> Hi, How can you subscribe to this service? There is no web page Dan> available to do it. I emailed them at [EMAIL PROTECTED], as per one of the pages on their web site. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold & Read error on sound device
You didn't mention the distro you are using. I'm wondering if you are using one of the distros that leans towards the alsa drivers. If so, then chan_oss would have problems. On Sun, 2003-07-20 at 19:16, Stuart Hirst wrote: > I am having a problem getting music on hold working one of my servers. > I have had this working on a PII 400 just fine but decided to upgrade > my Asterisk server to a PIV 1.5ghz. > > I have installed mpg123 which seems to be working fine but when I > start *, I get the following error message at the CLI prompt when I > start *: > > WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error > on sound device: Resource temporarily unavailable > > I have checked that the sound card works by loading X11 and running > sound tests which is fine. I have used "lsof /dev/dsp" to see if > another application or server is controlling the sound device and > without * running, nothing is reported. With * running lsof reports > that * has the device. Voicemail works fine. > > When I put a call on hold the CLI shows moh starting but nothing is > played. No errors are reported whilst starting moh. > > I have been trying lots of different things for hours now without > success. > > Anyone got any pointers ? > > Rgds, > > Stuart -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst wrote: When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? What does your musiconhold.conf file look like? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Summary of VoIP options for Asterisk and request for more?
Hello! Well, so much for mailing me off-list: not a single person did! In other words, you've already seen the results of my request: The options are: Nufone.net Cost: 2.9 cents/min for both outgoing long distance and incoming 800 calls. Service is pre-paid. Advantages: Extremely satisfied customers. IAX support. Disadvantages: One of the worst websites I've ever seen for a company. (Well, at least it doesn't use flash...) However, customers say that the customer side of the website is very good. www.global-gateway.net Cost: between 2.55 and 1.9 cents/min for calls (according to a customer) Advantages: I don't know. I could find out nothing about this company. Their website didn't work for me at all in either Mozilla or IE: All links merely pointed to "#"... Disadvantages: A completely broken website, no US telephone numbers. Seeing as only one person mentioned Global Gateway, and several raved about Nufone, I guess that's the direction I'm going to look. Nufone's service seems more geared to business use: (800) number for incoming, etc. I was more looking for something for home: a local telephone number, a number of minutes for a small monthly cost, etc. I'm going to have to review my home usage to see if it makes sense. However, my use of VoIP at home is in preparation for using it at the office, so maybe it's the best way to go... Are there any other options that people are using? IAX termination, while nice, is not required: SIP would work too. I'd prefer to stay away from H.323, but if you're using Asterisk to talk H.323 with a VoIP provider, I'd love to hear about it... Thank you very much for the responses. I really appreciate your help. Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold & Read error on sound device
Title: Message I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable I have checked that the sound card works by loading X11 and running sound tests which is fine. I have used "lsof /dev/dsp" to see if another application or server is controlling the sound device and without * running, nothing is reported. With * running lsof reports that * has the device. Voicemail works fine. When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? Rgds, Stuart
[Asterisk-Users] DTMF crashes chan_capi
Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set "softdtmf=1" in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection? The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP client is SJPhone (G711) on Windows connected to * via a 100Mbit switched LAN. PSTN connection is using latest chan_capi on an AVM B1 PCI v4 card (a steal at £1.20 on ebay :)). I can call the * box using the SIP client and interact with the voicemail app with no problems using in-band DTMF. I can also call in from the PSTN through the capi interface and interact with the IVR menu with no problems. Finally I can bridge the CAPI and SIP channels and hear DTMF digits entered on the PSTN phone with no problems (they are also detected and displayed on the console). However when the CAPI and SIP channels are bridged, entering more than a couple of DTMF digits into the _SIP_ client appears to crash the channel: neither party gets disconnected, but there is no longer any audio in either direction and new calls (inbound or outbound) trying to use the CAPI channel fail. Once locked if I enter "capi info" in the * console it return nothing and trying to autocomplete capi commands e.g. "capi [TAB]" just locks the console up. Entering capiinfo and lsmod at the command prompt suggests the driver is ok. The only way of getting it working again is to restart *. When I switch to softdtmf, everything seems to work fine, but I noticed that even though DTMF signalling works fine on the IVR menu, once the call is bridged DTMF digits entered on the PSTN phone are not displayed on the console like before. Jamie Neil Versado I.T. Services Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
This is generally indicates a problem with the licensing process (which is severely flawed and full of bugs) on your server... Did you make it through the registration process OK? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Sunday, July 20, 2003 12:18 AM To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk crashes when trying to load G.729 module. Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 => /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) => /lib/libc.so.6 libc.so.6 (GLIBC_2.2) => /lib/libc.so.6 libc.so.6 (GLIBC_2.1) => /lib/libc.so.6 libc.so.6 (GLIBC_2.0) => /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) => /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) => /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) => /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Is it stable enought? I mean around 30-40 Incoming SIP connections. Or i must trash the cisco and put Asteriisk/Digium/speex box? Jeremy McNamara wrote: > You have to run a console with the G.729 due to the voice age library > lameness. We run safe_asterisk with a TTY and it seems to be fine. > > > Jeremy McNamara > > > > [EMAIL PROTECTED] wrote: > > >>Try launching asterisk like this: >> >>screen -d -m asterisk -vvvcn >> >>Aparently there is some bug in the codec. >> >>- Justin >> >> >>On Sun, 20 Jul 2003, Anton Tinchev wrote: >> >> >> >> >>>Before few days i bought few g.729 licenses. >>>When i try to load the codec, asterisk crahses. >>>I tried with and without oh323 module, same result: >>>-- >>>Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable >>>to initialize va stuff: -1 >>>-- >>> >>>Here the ldd result: >>>-- >>>[EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so >>> libc.so.6 => /lib/libc.so.6 (0x40039000) >>> /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x8000) >>> >>> Version information: >>> /usr/lib/asterisk/modules/codec_g729b.so: >>> libc.so.6 (GLIBC_2.1.3) => /lib/libc.so.6 >>> libc.so.6 (GLIBC_2.2) => /lib/libc.so.6 >>> libc.so.6 (GLIBC_2.1) => /lib/libc.so.6 >>> libc.so.6 (GLIBC_2.0) => /lib/libc.so.6 >>> /lib/libc.so.6: >>> ld-linux.so.2 (GLIBC_2.1) => /lib/ld-linux.so.2 >>> ld-linux.so.2 (GLIBC_2.0) => /lib/ld-linux.so.2 >>> ld-linux.so.2 (GLIBC_PRIVATE) => /lib/ld-linux.so.2 >>>--- >>> >>>___ >>>Asterisk-Users mailing list >>>[EMAIL PROTECTED] >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Self parked but avaliable
Would this work with SIP / H323 phones?? - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 21, 2003 1:11 AM Subject: Re: [Asterisk-Users] Self parked but avaliable > On Sun, 2003-07-20 at 06:43, [EMAIL PROTECTED] wrote: > > Is there any way I can define an extension that I can call which will park my > > call so that I can listen to hold music over the speaker phone, but then if a > > real call comes in for me, it prompts me to press a key to accept the call, and > > if I do then it takes me out of parking can connects me to the incoming call? > > If you have call waiting set up, you could just define an extension that > is your music list. Then you dial your music extension, if a call comes > for you, you will hear a beep. During this beep you should get the CID > spill and be able to decide if you want to answer the line. If you want > to answer the line, hangup and the phone should ring with the incoming > call. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
On Sun, 20 Jul 2003, Linus Surguy wrote: > Hi all, > > We're currently running a PSTN -> SIP gateway with Asterisk. We also run > IAX/SIP -> PSTN. > > We have performed a test where the call is routed > > UK PSTN -> Digium E1 card -> Asterisk GW -> SIP G.711 -> FWD -> X-Ten > softphone > > There is no echo at the softphone end, but severe echo on the PSTN side. > > We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Best" VoIP provider for Asterisk?
Send email to [EMAIL PROTECTED] I did this late Friday afternoon and Jeremy had me set up in very short order. (I gave him wrong contact info Friday but he IM'd me the needed account info Saturday morning.) I've got problems with my own IP connection but aside from that the service just works. Once I resolve that I'll be able to truly vouch for the line quality. I've got an 800 number for incoming calls and I can route long distance calls out to NuFone. Both directions is $0.029/minute. They are still building their web based account management tools but I saw a preview and they look pretty nice. It's a prepay service and there doesn't seem to be an account set up fee right now so it's easy to get set up and try it out--that's what I'm doing. For what it's worth, I didn't do any investigation of alternatives. Good customer service, like NuFone appears to be in the business of, is usually worth a lot more than maybe getting the lowest rate. -reed At 09:20 AM 7/20/2003 +0300, you wrote: Hi, How can you subscribe to this service? There is no web page available to do it. Thanks, Dan - Original Message - From: "Erik Anderson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 19, 2003 6:07 PM Subject: RE: [Asterisk-Users] "Best" VoIP provider for Asterisk? > Agreed. Jeremy McNamara of Nufone.net is the top dog in Asterisk VOIP and > long distance. > > His systems and network are the most stable I have ever seen. It is all ran > out of the same facilities as the TOP long distance providers. All fiber, > all stable, 3x and 4x redundancy. > > He has done some amazing things. > > Erik > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED]]On Behalf Of James H. > > Cloos Jr. > > Sent: Saturday, July 19, 2003 5:47 AM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] "Best" VoIP provider for Asterisk? > > > > > > > "Marcus" == Marcus Adolfsson > > <[EMAIL PROTECTED]> writes: > > > > Marcus> Nufone.net is the best VoIP provider for Asterisk > > Marcus> integration. They offer IAX termination, 2.9 cents outgoing > > Marcus> long-distance and incoming 800. We use them at our office for > > Marcus> all phone calls. > > > > I second this. But note they are now at 2.0 cents for calls to US and > > Canada. They change the same per minute for incoming calls on the 800 > > numbers. > > > > They are responsive, competent; simply great to work with. > > > > Highly recommended. > > > > -JimC > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Self parked but avaliable
Maybe the agent stuff? Mark On Sun, 20 Jul 2003 [EMAIL PROTECTED] wrote: > Is there any way I can define an extension that I can call which will park my > call so that I can listen to hold music over the speaker phone, but then if a > real call comes in for me, it prompts me to press a key to accept the call, and > if I do then it takes me out of parking can connects me to the incoming call? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No audio in Messenger
Yo, I'm trying to get Asterisk working with Messenger 4.7. After skimming through the list-archives, I've got it to register to my Asterisk-box and can make calls. Unfortunately, there's no audio from the Messenger- side of the call to the other caller. I can hear the caller in Messenger, though. It doesn't appear to be a Messenger or network-problem, as I can talk to FWD from it just fine. This same problem also shows up on the PSTN to FWD-gateway I just set up. If the other end of the call uses Messenger, there won't be audio from it. It works fine with X Lite, tried from the same machine. I've tried everything from using different CODECs to "canreinvite=no", "nat=yes", "insecure=yes", etc, without much luck. Does anyone have an idea or got it working? I'm using the latest CVS for all Asterisk-stuff. Thanks in advance! Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'Echo' - I'm sure a common topic
Hi all, We're currently running a PSTN -> SIP gateway with Asterisk. We also run IAX/SIP -> PSTN. We have performed a test where the call is routed UK PSTN -> Digium E1 card -> Asterisk GW -> SIP G.711 -> FWD -> X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test BRI PBX -> AVM Fritz CAPI -> Asterisk -> IAX -> Asterisk GW -> E1 -> UK PSTN Once again, no echo at the BRI side, but some echo at the UK PSTN side. We do have echocancel=yes / echocancel = 128 on the Asterisk machine with the E1 card. Is there any other options we can turn on / look at to test this? Also, are their any 'README's which document the best usage of the options in (for example) the zaptel makefile for MMX and more agressive echo cancel etc. We havnt tried these yet as we were unsure of any caveats? Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
You have to run a console with the G.729 due to the voice age library lameness. We run safe_asterisk with a TTY and it seems to be fine. Jeremy McNamara [EMAIL PROTECTED] wrote: Try launching asterisk like this: screen -d -m asterisk -vvvcn Aparently there is some bug in the codec. - Justin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 => /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) => /lib/libc.so.6 libc.so.6 (GLIBC_2.2) => /lib/libc.so.6 libc.so.6 (GLIBC_2.1) => /lib/libc.so.6 libc.so.6 (GLIBC_2.0) => /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) => /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) => /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) => /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes when trying to load G.729module.
Try launching asterisk like this: screen -d -m asterisk -vvvcn Aparently there is some bug in the codec. - Justin On Sun, 20 Jul 2003, Anton Tinchev wrote: > Before few days i bought few g.729 licenses. > When i try to load the codec, asterisk crahses. > I tried with and without oh323 module, same result: > -- > Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable > to initialize va stuff: -1 > -- > > Here the ldd result: > -- > [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so > libc.so.6 => /lib/libc.so.6 (0x40039000) > /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x8000) > > Version information: > /usr/lib/asterisk/modules/codec_g729b.so: > libc.so.6 (GLIBC_2.1.3) => /lib/libc.so.6 > libc.so.6 (GLIBC_2.2) => /lib/libc.so.6 > libc.so.6 (GLIBC_2.1) => /lib/libc.so.6 > libc.so.6 (GLIBC_2.0) => /lib/libc.so.6 > /lib/libc.so.6: > ld-linux.so.2 (GLIBC_2.1) => /lib/ld-linux.so.2 > ld-linux.so.2 (GLIBC_2.0) => /lib/ld-linux.so.2 > ld-linux.so.2 (GLIBC_PRIVATE) => /lib/ld-linux.so.2 > --- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Best" VoIP provider for Asterisk?
hi all sorry, this price was wrong www.global-gateway.net does 2.55c/min for end users and down to 1.921 for wholesale roy On Sun, 2003-07-20 at 17:46, Roy Sigurd Karlsbakk wrote: > We're using http://www.global-gateway.net/ > > I've compared their pricing with nufone.com, and for what I can see, > they're quite a bit below (.us @ 21c/min). They do not, however, have > .us number termination. Their website sucks but the voip works. we have > an IAX2 trunk over to their .uk site. > > roy > > > On Fri, 2003-07-18 at 17:54, [EMAIL PROTECTED] wrote: > > > > > > > > > > Hello! > > > > I would like to get connected with a VoIP provider for home. At some > > point, I'm sure I will be connecting to it via an Asterisk box, but for > > now, I will be using whatever hardware they provide. > > > > What recomendations do you in the Asterisk community have for a reliable > > VoIP service that will hopefully interoperate with Asterisk? A company > > that is actually willing to work with an Asterisk user would be > > outstanding. They don't have to support the Asterisk box, but not hang up > > on me when I say that I'm using it would be wonderful. > > > > The companies I have researched are Vonage, iConnectHere and Packet8. I'm > > sure there are others, though. > > > > If you could reply off-list, I will be happy to type up a summary of the > > results and post it on-list later. > > > > Thank you! > > > > Tim Massey > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Best" VoIP provider for Asterisk?
We're using http://www.global-gateway.net/ I've compared their pricing with nufone.com, and for what I can see, they're quite a bit below (.us @ 21c/min). They do not, however, have .us number termination. Their website sucks but the voip works. we have an IAX2 trunk over to their .uk site. roy On Fri, 2003-07-18 at 17:54, [EMAIL PROTECTED] wrote: > > > > > Hello! > > I would like to get connected with a VoIP provider for home. At some > point, I'm sure I will be connecting to it via an Asterisk box, but for > now, I will be using whatever hardware they provide. > > What recomendations do you in the Asterisk community have for a reliable > VoIP service that will hopefully interoperate with Asterisk? A company > that is actually willing to work with an Asterisk user would be > outstanding. They don't have to support the Asterisk box, but not hang up > on me when I say that I'm using it would be wonderful. > > The companies I have researched are Vonage, iConnectHere and Packet8. I'm > sure there are others, though. > > If you could reply off-list, I will be happy to type up a summary of the > results and post it on-list later. > > Thank you! > > Tim Massey > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323
You have something drasticly wrong somewhere... 64 is SLINEAR and chan_h323 does nothing with SLINEAR frames. 1 is G.723.1 4 is G.711 u-law check your config...ensure your allow'ing the proper codec's. Jeremy McNamara [EMAIL PROTECTED] wrote: Having problems to connect another device using chan_h323. When G723.1 or G711: log says: NOTICE[15376]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 64 NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 4 to 1 WARNING[15376]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 4, while native formats is 1 (read/write = 64/4) WARNING[15376]: File app_dial.c, Line 299 (wait_for_answer): Unable to forward voice == No one is available to answer at this time But it works using chan_oh323. I appreciate any help. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Self parked but avaliable
On Sun, 2003-07-20 at 06:43, [EMAIL PROTECTED] wrote: > Is there any way I can define an extension that I can call which will park my > call so that I can listen to hold music over the speaker phone, but then if a > real call comes in for me, it prompts me to press a key to accept the call, and > if I do then it takes me out of parking can connects me to the incoming call? If you have call waiting set up, you could just define an extension that is your music list. Then you dial your music extension, if a call comes for you, you will hear a beep. During this beep you should get the CID spill and be able to decide if you want to answer the line. If you want to answer the line, hangup and the phone should ring with the incoming call. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Self parked but avaliable
Is there any way I can define an extension that I can call which will park my call so that I can listen to hold music over the speaker phone, but then if a real call comes in for me, it prompts me to press a key to accept the call, and if I do then it takes me out of parking can connects me to the incoming call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323
Having problems to connect another device using chan_h323. When G723.1 or G711: log says: NOTICE[15376]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 64 NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 4 to 1 WARNING[15376]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 4, while native formats is 1 (read/write = 64/4) WARNING[15376]: File app_dial.c, Line 299 (wait_for_answer): Unable to forward voice == No one is available to answer at this time But it works using chan_oh323. I appreciate any help. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
