Re: [Asterisk-Users] realtime excessive database queries
snacktime wrote: Personally I would rather see realtime load everything into memory and not go to the database unless something has changed or you reload. How would this be different than just storing the configuration statically in the database, and reading it at startup and at subsequent reloads? The primary deal with realtime, as I understand it, is that it *doesn't* keep the extensions tree (or whichever) in memory. Even the caching that's implemented for ARA right now takes it a bit away from that. Not trying to jab at you here, but try to understand the exact differentiation of function. Thx. B. ___ 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] MusicOnHold probelms
chawki hammoud wrote: --- Christian Wengel <[EMAIL PROTECTED]> wrote: Hi! Could you post your musiconhold.conf and modules.conf, please? MfG Christian This is the modules.conf file: [modules] autoload=yes noload => pbx_gtkconsole.so noload => pbx_kdeconsole.so load => cdr_addon_mysql.so ; noload => app_intercom.so ; load => chan_modem.so load => res_musiconhold.so ; noload => chan_alsa.so ; [global] chan_modem.so=yes And the musiconhold file: [classes] default => quietmp3:/var/lib/asterisk/mohmp3 [moh_files] Hi! Please try to add this line to your modules.conf in the [modules] section. noload => chan_oss.so Christian ___ 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 Project Consultant/Parner Wanted
Dear sir, I'm interested in your project. Can you tell more about it?? Regards. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, May 22, 2005 4:23 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk Project Consultant/Parner Wanted Hello All, How are you all doing today? Good I hope. I am sure that I have asked this question before, but recently lost my emails server and thus any replies that you may have sent me. We are working to get a small online VoIP service established and I am looking for someone who might like to partner on this project or possibly offer reasonable consulting services. We need someone to take the lead on the development of the Asterisk PBX server and site configuration to get the service set up and operating. Please send an email if interested. Have a good day, Lonnie ___ 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] realtime excessive database queries
> Rather than caching the data, which would remember the data past it's > useby date (which is never a fixed value, the useby date is when the > data in the DB is modified, which could be at anytime!), we should > simply read the entire extension in a single select, and cache that > answer for the life of the channel, or until it moves to another > extension. > > This will help assure that the extension is always in some consistent > state ie, it is either the old set of values, OR the new set, but > not going to start with the old set, get to priority 5, and suddenly > break because we changed things... I agree that's a good idea. However with a good cache mechanism you don't have to worry about getting old data. When you update the database you delete that item from the cache, which forces the client to get the new value from the database the next time it needs it. Or you tell the client somehow to reload the changed item. Personally I would rather see realtime load everything into memory and not go to the database unless something has changed or you reload. Then maybe add a new manager command like the following: RealtimeReload(context|all,extension|all) Which would trigger asterisk to reload from the database, locking access to the context/extension until the reload was complete. Another nice thing about this approach is that if your database (god forbid) goes down, asterisk can keep humming along. 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
[Asterisk-Users] spa-1001 with asterisk?
hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running [EMAIL PROTECTED] 1.0 can some one please tell me the steps to get the spa-1001 working for my friend, I will be passing the instructions to him. thanks for any help you can give. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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: Cisco 7940g Firmware load problems
Hello, We had the same problem. From my understanding - the Univeral Loader showed up somewhere around version 7. So if you have a 6 or less, or if you are coming from a different protocol - you first need to load the loader, which I think is POO3-07-4-00. After that loads, setup the configs to load POS3-07-4-00. The Universal loader will then know what to do with that file. I might have mixed up the order (P00 and P0S) - so just play with it - it will take you 5 min to update. Rafal Monday, May 23, 2005, 12:40:28 AM, you wrote: BG> While I was trying to upgrade cisco 7940 to sip it gives the same error Protocol Applicatation BG> Invalid because of lack of some necessary files such sip, sep,XMLDefault.CNF BG> BG> The files included in my tftp server are BG> BG> OS79XX BG> P0S3-07-4-00.sb2 BG> P003-07-4-00.sbn BG> P0S3-07-4-00 BG> P0S3-07-4-00.loads BG> P003-07-4-00 BG> Dialplan BG> SEP BG> SIPDefault BG> xmlDefault BG> CTLSEP.tlv BG> SIP BG> RINGLIST BG> From: Adam Collard [mailto:[EMAIL PROTECTED] On Behalf Of Adam Collard BG> Sent: Monday, May 23, 2005 7:54 AM BG> To: asterisk-users@lists.digium.com BG> Subject: Cisco 7940g Firmware load problems BG> BG> I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in BG> Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that BG> are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can BG> do to fix both these problems. I am running [EMAIL PROTECTED] version 1.0. I have to of Cisco 7940g's working BG> perfectly on my server right now. I can be reached at (800) 757-5669 x4861 or [EMAIL PROTECTED] I BG> need these working right ASAP. BG> BG> Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini BG> hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar BG> geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi BG> kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza BG> yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida BG> belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup BG> Important note : This e-mail transmission is intended only for the use of the individual or entity to BG> which it is addressed, and may contain information that is privileged, confidential and that may not be BG> made public by law or agreement. If the recipient of this message is not the intended recipient or entity, BG> you are hereby notified that any further dissemination, distribution or copying of this information is BG> strictly prohibited. If you have received this communication in error, please notify us immediately by BG> telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup -- Best regards, Rafalmailto:[EMAIL PROTECTED] Friendly Solutions Corp. 213 S. Wille St. Mount Prospect, IL 60056-3120 office: (773) 957-7800 mobile: (847) 312-4567 www.friendlysol.com mailto:[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] SIP Stress Test
Hello, in January, there was announced a graphical asterisk load tester on this list. Can anyone tell me, where I can find this tool? Regards, Henry ___ 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] IAX2 native phone
Hello, Anybody can recommend a IAX2-native IP phone? Preferably buyable online.. TIA. ___ 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] IAXTEl down
> Figures... So... Everybody went to FWD :) ? It mostly works, does IAX, so, yeah. ___ 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] Looking for people to test calls
We are looking for people to help test our termination services before going into production. The more problems, the better, we want to debug everything before we start selling. In exchange for your help, you will get a discount on services once we are fully operational. If you are interested in testing, please reply off-list. If you are in Mexico, please disregard. ___ 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: Cisco 7940g Firmware load problems
Title: Asterisk-Users Digest, Vol 10, Issue 174 While I was trying to upgrade cisco 7940 to sip it gives the same error Protocol Applicatation Invalid because of lack of some necessary files such sip, sep,XMLDefault.CNF The files included in my tftp server are OS79XX P0S3-07-4-00.sb2 P003-07-4-00.sbn P0S3-07-4-00 P0S3-07-4-00.loads P003-07-4-00 Dialplan SEP SIPDefault xmlDefault CTLSEP.tlv SIP RINGLIST From: Adam Collard [mailto:[EMAIL PROTECTED] On Behalf Of Adam Collard Sent: Monday, May 23, 2005 7:54 AM To: asterisk-users@lists.digium.com Subject: Cisco 7940g Firmware load problems I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running [EMAIL PROTECTED] version 1.0. I have to of Cisco 7940g's working perfectly on my server right now. I can be reached at (800) 757-5669 x4861 or [EMAIL PROTECTED]. I need these working right ASAP. Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ 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] Which H.323 for Stable?
I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323 experience offer any meaningful advice as to which way I _must_ proceed? This is for a live, busy, deployed environment. H.323 will be used to connect to an upstream provider (possibly CISCO gear?). Thanks in advance. ___ 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] realtime excessive database queries
> we should > simply read the entire extension in a single select, and cache that > answer for the life of the channel, or until it moves to another > extension. That sounds like a good idea. -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] Hangup Issues on TDM40B FXO Australia
Thanks Terry noticed [EMAIL PROTECTED] 0.7 will try version 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Monday, 23 May 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Monday, 23 May 2005 12:49 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia > > Afternoon all, > > After doing some test on my asterisk box I can successfully receive > calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network > > Dial out from a sip phone is also not an issue, all calls connect and > terminate normally. > > If I call the Asterisk PBX say from PSTN in Zap1-1 and out through > ZAP2-2 back to the PSTN (after entering the correct pin off > course) the card does not appear to detect the hang-up, I then have to > issues a soft hang-up to close the call, I presume this indicates the > card is configured to receive the correct hangup signal > > I have tried enabling callprogress, busydetect and a few settings on > the busycount but to no success > > I've also tried LS and KS signalling > > Does anyone else have any suggestions to get this to work with > Australia's Telstra? > > > > Regards > > Haydn > > > > > > > -- > -- > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. > If you have received this email in error please notify the originator > of the message. This footer also confirms that this email message has > been scanned for the presence of computer viruses. > > Any views expressed in this message are those of the individual > sender, except where the sender specifies and with authority, states > them to be the views of LMC. > > -- > -- > > ___ > 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 This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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] Hangup Issues on TDM40B FXO Australia
Thanks for the response all I'm Currently running version 1.0.7 of asterisk, Key configuration lines as below, for those who have it working is there anything that stands out as incorrect? * Zaptel.conf fxsls=1-4 loadzone=au defaultzone=au *** Zapata.conf signalling=fxs_ls switchtype=national *** zapata-channels.conf signalling=fxs_ls *** zapata_additional.conf ;;[EXT] signalling=fxo_ls echotraining=400 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=yes busydetect=yes busycount=3 channel=>1 *** -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Monday, 23 May 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Monday, 23 May 2005 12:49 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia > > Afternoon all, > > After doing some test on my asterisk box I can successfully receive > calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network > > Dial out from a sip phone is also not an issue, all calls connect and > terminate normally. > > If I call the Asterisk PBX say from PSTN in Zap1-1 and out through > ZAP2-2 back to the PSTN (after entering the correct pin off > course) the card does not appear to detect the hang-up, I then have to > issues a soft hang-up to close the call, I presume this indicates the > card is configured to receive the correct hangup signal > > I have tried enabling callprogress, busydetect and a few settings on > the busycount but to no success > > I've also tried LS and KS signalling > > Does anyone else have any suggestions to get this to work with > Australia's Telstra? > > > > Regards > > Haydn > > > > > > > -- > -- > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. > If you have received this email in error please notify the originator > of the message. This footer also confirms that this email message has > been scanned for the presence of computer viruses. > > Any views expressed in this message are those of the individual > sender, except where the sender specifies and with authority, states > them to be the views of LMC. > > -- > -- > > ___ > 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 This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/
[Asterisk-Users] Cisco 7940g Firmware load problems
I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running [EMAIL PROTECTED] version 1.0. I have to of Cisco 7940g's working perfectly on my server right now. I can be reached at (800) 757-5669 x4861 or [EMAIL PROTECTED] I need these working right ASAP. <>___ 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] Using patch -p0
That patch is very small, so if you need to you could manually apply the patch. All it adds is callerId to the "meetme list confno" command. It is based on 1.0.7, and I did apply it to a clean tree to verify it, but I am also the first to admit that I am new to using diff/patch, so I may have done it wrong. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Flash Love Sent: Sunday, May 22, 2005 2:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Using patch -p0 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 to forward a call to mobile?
i have an account with BV on my asterisk, how to forward a unanswered incoming call to my mobile phone ( when there is no one to answer the incoming call after 3 rings) ? ___ 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] IAX losing registration
Joel Duffield wrote: The problem is still occuring. it happens even if I register with myself, it works for some time and then just dies. The qualify still shows up as 65ms on the outside server, but the registry just says "Request Sent". and a reload doesn't help only restart. Are you registering against a hostname or IP? Try changing to an IP if possible. Is it maybe caching DNS somewhere along the line for that Host (and maybe the IP has changed)? Strange that reload does not work as I thought that it cleared the dns cache (as far as Asterisk is concerned). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Hangup Issues on TDM40B FXO Australia
I had a similar issue both with the X100P clones and TDM400. Both were fixed by enabling AU zone and the busydetect functions. Don't forget a full asterisk reload needs to take place after changing Zap conf files, not just a soft-reload. Best way is to reboot the computer. Mike > I have a similar issue. > > I have 2 pstn lines and a phone plugged into my tdm400. > If I make a call to the outside using the phone, and the pstn number is > engaged, and I hang up, the line is not freed. I have been restarting > asterisk to get my external line back. > > This does not happen if I make the same call from my pc (using sj phone). > > Malcolm > > [EMAIL PROTECTED] wrote: > >>Afternoon all, >> >>After doing some test on my asterisk box I can successfully receive >>calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network >> >>Dial out from a sip phone is also not an issue, all calls connect and >>terminate normally. >> >>If I call the Asterisk PBX say from PSTN in Zap1-1 and out through >>ZAP2-2 back to the PSTN (after entering the correct pin off course) the >>card does not appear to detect the hang-up, I then have to issues a soft >>hang-up to close the call, >>I presume this indicates the card is configured to receive the correct >>hangup signal >> >>I have tried enabling callprogress, busydetect and a few settings on the >>busycount but to no success >> >>I've also tried LS and KS signalling >> >>Does anyone else have any suggestions to get this to work with >>Australia's Telstra? >> >> >> >>Regards >> >>Haydn >> >> >> >> >> >> >> >>This email and any files transmitted with it are confidential and >>intended solely for the use of the individual or entity to whom >>they are addressed. >>If you have received this email in error please notify the >>originator of the message. This footer also confirms that this >>email message has been scanned for the presence of computer viruses. >> >>Any views expressed in this message are those of the individual >>sender, except where the sender specifies and with authority, >>states them to be the views of LMC. >> >> >> >>___ >>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 >> >> >> >> >> > > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 > > ___ > 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] Hangup Issues on TDM40B FXO Australia
Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Monday, 23 May 2005 12:49 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia > > Afternoon all, > > After doing some test on my asterisk box I can successfully > receive calls to my Asterisk PBX to a SIP phone from The > Telstra PSTN network > > Dial out from a sip phone is also not an issue, all calls > connect and terminate normally. > > If I call the Asterisk PBX say from PSTN in Zap1-1 and out through > ZAP2-2 back to the PSTN (after entering the correct pin off > course) the card does not appear to detect the hang-up, I > then have to issues a soft hang-up to close the call, I > presume this indicates the card is configured to receive the > correct hangup signal > > I have tried enabling callprogress, busydetect and a few > settings on the busycount but to no success > > I've also tried LS and KS signalling > > Does anyone else have any suggestions to get this to work > with Australia's Telstra? > > > > Regards > > Haydn > > > > > > > -- > -- > This email and any files transmitted with it are confidential > and intended solely for the use of the individual or entity > to whom they are addressed. > If you have received this email in error please notify the > originator of the message. This footer also confirms that > this email message has been scanned for the presence of > computer viruses. > > Any views expressed in this message are those of the > individual sender, except where the sender specifies and with > authority, states them to be the views of LMC. > > -- > -- > > ___ > 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] Not answering/script.
On Sun, 2005-05-22 at 23:01 -0400, Ken D'Ambrosio wrote:
> I've got an Asterisk box at a client; last week, it just stopped answering
> the phone. Outbound calls still went, but inbound -- no dice. Asterisk
> didn't even acknowledge that the line was ringing. A reboot fixed it --
> though, clearly, I can't have them rebooting all the time.
> - Until I resolve the issue, how would I go about writing a script to call
> the customer from my own Asterisk box, and, if the call doesn't get
> answered after (say) 30 seconds, send me an e-mail saying that
> something's wrong? I'm not even sure where to start with something like
> this, but I have to imagine it's been done before.
How about a cron job to drop a .call file on your PBX which will go to a
context like this:
exten => s,1,Dial(ZAP/g1/,30)
exten => s,2,gotoif(${DIALSTATUS}="unanswered",10)
exten => s,3,noop(allok)
exten => s,4,Hangup
exten => s,10,noop(BAD)
exten => s,11,system(mail -s "Customer PBX XXX is not answering calls"
[EMAIL PROTECTED])
exten => s,12,hangup
You will need to look at priority 2, and fix that to work, I didn't look
at the usage/etc for that.
Regards,
Adam
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Re: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
I have a similar issue. I have 2 pstn lines and a phone plugged into my tdm400. If I make a call to the outside using the phone, and the pstn number is engaged, and I hang up, the line is not freed. I have been restarting asterisk to get my external line back. This does not happen if I make the same call from my pc (using sj phone). Malcolm [EMAIL PROTECTED] wrote: Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 ___ 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] Not answering/script.
I've got an Asterisk box at a client; last week, it just stopped answering the phone. Outbound calls still went, but inbound -- no dice. Asterisk didn't even acknowledge that the line was ringing. A reboot fixed it -- though, clearly, I can't have them rebooting all the time. So: - Should I be thinking hardware or software as the issue? - Until I resolve the issue, how would I go about writing a script to call the customer from my own Asterisk box, and, if the call doesn't get answered after (say) 30 seconds, send me an e-mail saying that something's wrong? I'm not even sure where to start with something like this, but I have to imagine it's been done before. Thanks! Ken D'Ambrosio ___ 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 with vonage linksys adapter? {Scanned}
No, but I got a SPA-2000 from ebay for $40 bucks. David On Sun, 2005-05-22 at 20:57 -0500, Matthew Boehm wrote: > Short Answer: No > > For the long answer: google.com > > -Matthew > > > > From: hank smith <[EMAIL PROTECTED]> > > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Date: Sun, 22 May 2005 13:31:12 -0700 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: [Asterisk-Users] asterisk with vonage linksys adapter? > > > > hello do you know if vonage unlocks there linksys adapter to use with other > > providers? I want to use my ixisting vonage adapter with asterisk and cancil > > my vonage service. > > thanks > > hank > > > > email: > > [EMAIL PROTECTED] > > gmail: > > [EMAIL PROTECTED] > > msn messenger: > > [EMAIL PROTECTED] > > aim: > > hanksmith5 > > skype: > > hanksmith5 > > ___ > > 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] Hangup Issues on TDM40B FXO Australia
Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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] realtime excessive database queries
On Sun, 2005-05-22 at 18:44 -0700, snacktime wrote: > On 5/22/05, Preston Garrison <[EMAIL PROTECTED]> wrote: > The biggest problem is several routines > > would need to be modified to pass around the data. It was just easier > > to put hooks in all the functions for the database. Writing your own > > database interface and caching the queries could do it as well. > > I was thinking about using postgresql with odbc since postgresql has > an interface to memcached which we use a lot anyways. And with > postgresql I could just use a few rules to rewrite the queries from > asterisk and not have to touch the asterisk odbc code. This also > makes sense for us because we are already using postgresql with > memcached for a bunch of other stuff. Rather than caching the data, which would remember the data past it's useby date (which is never a fixed value, the useby date is when the data in the DB is modified, which could be at anytime!), we should simply read the entire extension in a single select, and cache that answer for the life of the channel, or until it moves to another extension. This will help assure that the extension is always in some consistent state ie, it is either the old set of values, OR the new set, but not going to start with the old set, get to priority 5, and suddenly break because we changed things... Just my 0.02c worth, or less, since I haven't even read the code ___ 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 with vonage linksys adapter?
Short Answer: No For the long answer: google.com -Matthew > From: hank smith <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Sun, 22 May 2005 13:31:12 -0700 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] asterisk with vonage linksys adapter? > > hello do you know if vonage unlocks there linksys adapter to use with other > providers? I want to use my ixisting vonage adapter with asterisk and cancil > my vonage service. > thanks > hank > > email: > [EMAIL PROTECTED] > gmail: > [EMAIL PROTECTED] > msn messenger: > [EMAIL PROTECTED] > aim: > hanksmith5 > skype: > hanksmith5 > ___ > 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] realtime excessive database queries
On 5/22/05, Preston Garrison <[EMAIL PROTECTED]> wrote: > Why not add the code to do so? I'm not the best C programmer or I would. I can see what needs to be done and could put all the logic together, but the actual coding would take me a while. The biggest problem is several routines > would need to be modified to pass around the data. It was just easier > to put hooks in all the functions for the database. Writing your own > database interface and caching the queries could do it as well. I was thinking about using postgresql with odbc since postgresql has an interface to memcached which we use a lot anyways. And with postgresql I could just use a few rules to rewrite the queries from asterisk and not have to touch the asterisk odbc code. This also makes sense for us because we are already using postgresql with memcached for a bunch of other stuff. 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] more than one company hosting their PBX on thesame machine?
Chris Mason (Lists) wrote: This interests me as I find I have several customers who cannot afford a complete hosting computer but need some extensions online. Domains are of no importance, I would think you cold give them each a block of extensions, i.e., 1xx, 2xx, they would know there were other companies on the machine but why would they care? What would be the issues? IVRs? I think voicemail has context for multi company hosting. I don't think that such a restriction has to be made, since you can have each phone and each line into a different contxt. To call from line 1 to extension 501 could be different than from line 2. Same from the phones. Sure, you need to make than for each phone the short number 501 and an extra large real one, so that you can distinguish them in the CDRs, ... bye Ronald Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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] realtime excessive database queries
Why not add the code to do so? The biggest problem is several routines would need to be modified to pass around the data. It was just easier to put hooks in all the functions for the database. Writing your own database interface and caching the queries could do it as well. We eventually dumped the idea of using an external database period, and ended up putting our own hash based system in place. Preston Garrison direct: 877-748-4142 fax: 310-774-3901 cell: 623-748-4140 -Original Message- From: Matthew Boehm <[EMAIL PROTECTED]> To: snacktime <[EMAIL PROTECTED]>; Asterisk Users Sent: Sun, 22 May 2005 12:27:06 -0500 Subject: Re: [Asterisk-Users] realtime excessive database queries The queries you speak of are not native to realtime. They are part of pbx as a whole. And yes, the core programmers are aware of it and yes, they have admitted they don't like it. > Why not just load all the extensions into memory, and then have a > mechanism to mark an extension as stale and only then reload from the > database? Seems that with a basic caching mechanism you get all the > benefits of realtime without the downside of taking a performance hit > like it does now when having to go to the database so much. > > If I'm reading the code right the extensions are all in a linked list. > So it seems you could just add another item to the list to hold the > state of the extension, and when accessing the first item in the list > check the state. If it's stale and realtime is in use, update the > extension from the database and restart at the top of the list. > > Chris _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] 2 Asterisk boxes sharing dial plans. {Scanned}
Well not what I was thinking. I would like to share the outbound trunks. One server needs an extra line it could use the other server. Thanks, David On Sun, 2005-05-22 at 16:42 -0400, Race Vanderdecken wrote: > Could you set up an NFS directory that is shared between the servers? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of David Shaw > Sent: Sunday, May 22, 2005 9:50 AM > To: Asterisk Users Mailing List > Subject: [Asterisk-Users] 2 Asterisk boxes sharing dial plans. > > Hello All, I have two asterisk boxes. 1 for home and 1 for work/ham > radio club. I have 2 SIP trunks on each server. What is the best way to > share the trunks? > > Thanks, David > > PS FWDOUT is great!!! > > ___ > 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] IAXTEl down
Figures... So... Everybody went to FWD :) ? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Domingo, 22 de Mayo de 2005 08:23 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] IAXTEl down | |> Is iaxtel down? |> |> Ive been getting this: |> May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: |> Auto-congesting call due to slow response |> -- IAX2/Iaxtel-12 is circuit-busy |> -- Hungup 'IAX2/Iaxtel-12' |> |> is it down or am I doing something wrong? | |Its been doing that for months. No one is actually maintaining |the site. | | |___ |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] IAXTEl down
> Is iaxtel down? > > Ive been getting this: > May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: > Auto-congesting call due to slow response > -- IAX2/Iaxtel-12 is circuit-busy > -- Hungup 'IAX2/Iaxtel-12' > > is it down or am I doing something wrong? Its been doing that for months. No one is actually maintaining the site. ___ 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] CallerID
Let mek now what you need Florian and Ill send it offlist. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 03:56 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |Citeren Anton Krall <[EMAIL PROTECTED]>: | |> Seems to me Im been displayed both... How can I control it? | |No way to know that without more in-depth knowledge about your |configuration (i.e.dialplan, what channel have you configured |in asttapi etc.) | |Florian |___ |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] Upgrade cause's no Audio on IAX
Further to this I have done a full reinstall of everything including ztdummy and asterisk to the CVS version downloaded yesterday. I get a Loud Buzzing when the line answers now and leaving a voicemail mesg just leaves blank :( So I am thinking as it appears that the ATA works correctly with g729a that it would be either IAX2 problems or iblc codec problems :( Has anyone got any advice? Thanks David David Uzzell wrote: Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction. Does anyone know what would have changed to cause this or what I would need to do to look at solving the issue ? I am now offline :( and for some reason rolling back to the older version now does not want to run :( My IAX conf [general] tos=lowdelay jitterbuffer=no disallow=all allow=speex allow=ilbc allow=gsm register => user:[EMAIL PROTECTED] [guest] type=user context=default auth=none [2347] type=friend username=user secret=password auth=md5 host=gw1.austechpartnerships.com context=default trunk=yes qualify=3000 disallow=all allow=ilbc Thanks 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 ___ 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] more than one company hosting their PBX on the samemachine?
On Sun, 2005-05-22 at 18:34 -0400, Chris Mason (Lists) wrote: > One potential problem wuold be CDRs, if they have access to them they would > see the other customers. I would think one could cutomize the CDR > applicastion to query only for the context for that company. > I dont see this as a problem for two reasons, in the cdr-csv there is a Master.csv that is generated for all calls, and each 'acountcode' has its own file named accountcode.csv. If using a database you can select based on that (or whatever else). So they would be able to get what they need. I would question giving people shell access to the box, that is too prone to problems, which means that you will need some other means to give them access, which inherently implies there is some processing. -- 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] more than one company hosting their PBX on the same machine?
On Sun, 2005-05-22 at 21:45 +, [EMAIL PROTECTED] wrote: > Did anyone try to have more than one asterisk installation on one machine? The bigger question is why ignore 'contexts' to run multiple instances? Contexts give you out of the box ability to route certain calls to certain places for independant IVR systems, independant voicemail, independant channel grouping, etc. You can vhost asterisk in this way quite easily without the extra overhead of extra instances. The only thing you have to be careful of is naming collisions (hint prefix or postfix names with some company identifier eg [Cust1-incoming]). Gotta be careful when doing this so that all 'goto' and similar are rewritten as well if the customer has the ability to create raw entries (if done via a tool, web or otherwise the tool could make sure this doesnt happen). -- 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] Allied Telesyn AT-VP504E and asterisk
Has anyone used one of these with an asterisk server? I am looking for a "cheap" FXS for my home server and spotted a couple of these on Ebay. One has only a couple of days to run and has so far got no bids! So, either it is rubbish or no-one has found it yet!! Any comments appreciated -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 20/05/2005 ___ 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] more than one company hosting their PBX on the samemachine?
One potential problem wuold be CDRs, if they have access to them they would see the other customers. I would think one could cutomize the CDR applicastion to query only for the context for that company. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Rod Bacon > Sent: Sunday, May 22, 2005 6:25 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] more than one company hosting their > PBX on the samemachine? > > Sigh... read the wiki. Search the lists. This has been > answered at least fifteen times. > > You don't need multiple instances of *, just set up your > dialplan properly. > > Hint: Contexts are the key. > > > ___ > 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] more than one company hosting their PBX on thesame machine?
This interests me as I find I have several customers who cannot afford a complete hosting computer but need some extensions online. Domains are of no importance, I would think you cold give them each a block of extensions, i.e., 1xx, 2xx, they would know there were other companies on the machine but why would they care? What would be the issues? IVRs? I think voicemail has context for multi company hosting. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] more than one company hosting their PBX on the same machine?
Sigh... read the wiki. Search the lists. This has been answered at least fifteen times. You don't need multiple instances of *, just set up your dialplan properly. Hint: Contexts are the key. ___ 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: Who knows where voicepulse has their asterisk servers?
I could have phrased the question better. If voicepulse, packet8, or vonage hops onto a tdm network (pstn) in one place more than any other (because they got a great rate on did and termination from a providers tdm network, so it's just easier for them to terminate a majority of their minutes in one carrier hotel than maintain their own gateways and contracts in many latas... where would that carrier hotel be, and who would be the clec handing them tdm access and did's there? A loaded question, but the answer's got to be useful for many of us here, right? ___ 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] MusicOnHold Loudness/Distortion
> For whatever reason, the music on hold is extremely distorted and loud. > It didn't used to be this way and I haven't changed anything, yet it > persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can > anyone help with this, or has anyone seen this? The mp3s play fine on > any computer and haven't changed since they did work. > Those wishing to hear for themselves, feel free to call extension 8800 > at the number/addresses below. > > Bryce Chidester It depends on what software you use for MOH. mpg123 is the default. Check the switches used by mpg123 and see if there is an option to lower the volume and then change it in musiconhold.conf. If I understand it correctly, the -g option is what you want. I use madplay. It has an option to lower the volume. Kevin ___ 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] Using patch -p0
I have googled for several hours and have read several threads but I have not found an answer yet. I have downloaded asterisk-1.0.7 and WebMeetMe-Gui. I have tried to use the diff file 'meetme-diff-cbmysql_1.txt' to add the changes needed for WebMeetMe-Gui. Using 'patch -p0 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more than one company hosting their PBX on the same machine?
Did anyone try to have more than one asterisk installation on one machine? Here's what i'm thinking, why can't amall and medium companies outsource their telephony service to their data provider? So the ISP could offer a service to its clients where it hosts the servers that run asterisk at their own premises which is usually a suitable environment for hosting servers(UPS's and very good network connectivity). Summary: Is it possible to have more than one domain(as in the extensions don't know each other) on the same machine? If it is, any recommended solutions? Many virtual machines and each one has one asterisk installation? One linux box that has several asterisk installations? What would be the hardware requirements for running such a setup? Regards, Akef ___ 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] MusicOnHold Loudness/Distortion
> For whatever reason, the music on hold is extremely distorted > and loud. > It didn't used to be this way and I haven't changed anything, yet it > persists. This is on all the channels we use (SIP, IAX2, Zap, > ALSA). Can > anyone help with this, or has anyone seen this? The mp3s play fine on > any computer and haven't changed since they did work. > Those wishing to hear for themselves, feel free to call extension 8800 > at the number/addresses below. > Bryce Your DTMF recognition seems screwed up. I can't get ex 8800, but I can get the MOH by dialling 80. Found that out by accidently misdialling the wrong extension. There's slight echo on your line too, and the voices sound "muddy" somehow. Can't help you with the dodgy MOH, sorry. Tom ___ 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] Help with follow me
Race Vanderdecken wrote:
A Rube Goldberg type solution is to send a text message to the cell
phone, reply to the message if you want the call to forward to your cell
phone.
You do have to keep the ZAP connection waiting for the SMS/text round
trip.
Race "they Tyrant" Vanderdecken
The way SMS/text works with some cell systems would make it Super Rube
Goldberg. Otherwise, I would be doing it already. It would provide a way
to do non-blind call transfer with caller ID name/number displayed.
With my cell provider there is a web page for sending text messages. I
looked at the html and then wrote a shell script that sends me text
messages. Sending is free. Replies are not.
#!/bin.sh
# Gnu GPL applies
# cheap and lazy approach
# use perl if you want to do anything fancy
wget -o sendpage.log -O /dev/null \
"http://${SOMEURL}/index.php?name=uscc&number=${SOMECELL}&message=testing99&Send=Send";
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RE: [Asterisk-Users] CallerID
Hi, Citeren Anton Krall <[EMAIL PROTECTED]>: > Seems to me Im been displayed both... How can I control it? No way to know that without more in-depth knowledge about your configuration (i.e.dialplan, what channel have you configured in asttapi etc.) Florian ___ 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] G729 codec
Gary Thanks for the info: do you know if the g729 codec for Asterisk is limited to only the G.729A version. Is there G.729 B available for Asterisk? - Original Message - From: Gary Lawrence To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Sunday, May 22, 2005 11:16 AM Subject: RE: [Asterisk-Users] G729 codec At the cli prompt type show codecs. In the right hand column it states G.729A. Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of toddSent: Sunday, May 22, 2005 12:55 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] G729 codec Hi allI have a question and hope it has not been answered before. I have searched the forums and mail but have not seen this answered conclusively.Does the G729 codec and licenses which digium sales for asterisk use g729 aor b or both; I have had a hard time getting a conclusive answer. If it does use g729b how could I show evidence to a client that it is b and not a?ThanksSincerelyTKG ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Digium and IPsando announces agenda for Astricon Europe - register now!
The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one that will be another milestone in Asterisk history. Find the information you need to register on http://www.astricon.net and register today! Some additional information, a mini-FAQ: * Hotel registrations is handled by our registration systems * You can register for dCAP testing in the registration system. We will contact you later for the time and day for the dCAP - that will be planned when we know the number of participants, since dCAP requires a lot of equipment. * There is still openings for tutorials. Send proposals to me. * There are still sponsorship oppurtunities. If you have any questions, don't hesitate to contact us on [EMAIL PROTECTED] See you in lovely Madrid! Best regards, /Olle and Steve --- Among the speakers: * Keynote by Mark Spencer, Digium * Ed Guy, Pulver: The architecture for FWDout * Paul Mahler, Signate: Scaling Asterisk * Nicolas Guidino: The Flash Operator Panel (FOP) * Izzy Gal, Xorcom: Threats and Opportunities for PBX Manufacturers and Telephony VARs in the Age of Asterisk * Rickard Lander: Implementing Voice over Wireless Mesh * Kristian Kielhofner: Introduction to ASTLinux * Nicholas Barnes: Selling Asterisk * Serge Kruppa, Virtual Contact center based on Asterisk * Femi Monehim: Using Asterisk for a carrier in Nigeria * Caleb Kow: Internationalization of Asterisk * David Troy: Developing Real Time Web-Telephony Applications with Asterisk * Nicholas Barnes: Selling Asterisk solutions Tutorials * Matt Fredrickson, Digium: The Zaptel and LibPRI architecture: An Overview * Matt Nicholsson, Digium: Extending Asterisk: The AMI and the AGI * Serge Kruppa: Building a carrier class hosted contact center platform with Asterisk * James Jones, Signate: Res_Perl: Perl embedded in your Asterisk! * Kindy Conley: The basics of telephony billing * Olle E. Johansson: Adventures with Asterisk and SIP * Keving P. Fleming, Digium: So you want to be an Asterisk developer? * ManxPower, Eric Wieling * Joachim (Zoa) Vanheuverzwijn: Asterisk security * Sergio Serrarno, Avanzada 7: Introduction to Asterisk (in spanish) ___ 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] 2 Asterisk boxes sharing dial plans.
Could you set up an NFS directory that is shared between the servers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Shaw Sent: Sunday, May 22, 2005 9:50 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] 2 Asterisk boxes sharing dial plans. Hello All, I have two asterisk boxes. 1 for home and 1 for work/ham radio club. I have 2 SIP trunks on each server. What is the best way to share the trunks? Thanks, David PS FWDOUT is great!!! ___ 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] Trouble using two Fritz ISDN cards in one machine
Hi, I'm heaving trouble using two Fritz ISDN cards in one machine. I followed "http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO"; and i think everything works great. cat /proc/capi/controller shows: 1 fcpci running fritz-pciA1 3.11-02 0xb400 5 2 f2pci running fritz2-pci A1 3.11-02 0xa400 10 The asterisk CLI shows: *CLI> capi info Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. Controller 1 works great, the problem is controller 2. I'm unable to recieve or dial-out over controller 2. My cat /etc/asterisk/capi.conf looks like this: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=49239XX1X (X to protect my privacy ;-)) incomingmsn=* controller=1 softdtmf=1 callgroup=1 context=extern devices=2 msn=49239XX1X (X to protect my privacy ;-)) incomingmsn=* controller=2 softdtmf=1 callgroup=1 context=extern devices=2 I'm using asterisk 1.0.7 with chan_capi 0.3.5. What am i missing /what am i doing wrong? Steven. ___ 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 with vonage linksys adapter?
hello do you know if vonage unlocks there linksys adapter to use with other providers? I want to use my ixisting vonage adapter with asterisk and cancil my vonage service. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] Help with follow me
A Rube Goldberg type solution is to send a text message to the cell phone, reply to the message if you want the call to forward to your cell phone. You do have to keep the ZAP connection waiting for the SMS/text round trip. Race "they Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Johnson Sent: Friday, May 20, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help with follow me I hope someone can help me with this. This is what I want to happen. Someone dials in and goes to my extension. First, the phone on my desk rings If there is not an answer, I would like to have the dialplan call my cell phone. If I answer my cell phone, speak the incomming number to me. I press one of the buttons on my cell phone to accept the call. If I don't answer, or I don't press the correct key on my cell, the call gets transfered into voicemail. In my searching I did see something like this is possible if I am dialing my cell phone from a Zap connection. Is there a way I can do something like this with my IAX connection??? Thanks Much Ben ___ 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] How can you keep agents logged in across a restart?
Yes, I have done what is called a Zombie list. I save the current registrations list every time a new registration comes in. When asterisk recycles I send a SIP message to everyone in the zombie list asking them to reregister with asterisk. Part of the SIP protocol. Mind you this only works with SIP. It is changes to the chan_sip.c code. Race "the tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Saturday, May 21, 2005 12:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How can you keep agents logged in across a restart? The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon. ___ 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] (another) cisco 7960 question
Use the Directory or Services to create a speed dial list. On 5/22/05, Nabeel Jafferali <[EMAIL PROTECTED]> wrote: > > My 7960 is configured for two lines, and I can turn the other appearance > > buttons into speed dials from the menus, but is there any way to program > > the speed dials in the SIP.conf file? > > You can not: http://tinyurl.com/az4fp > > -- > Nabeel Jafferali > X2 Networks > www.x2n.ca > T: 1.647.722.6900 >1.877.VOIP.X2N > F: 1.866.655.6698 > FWD: 46990 > > ___ > 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] MusicOnHold Loudness/Distortion
[Cross-posted and re-sent; it really sounds bad and needs resolution ASAP] For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below. Thank you, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305 ___ 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] Polycom IP600 Questions
1. How do you set the music on hold to work with asterisk. Right now when I place a call on hold the caller hears nothing. MOH works with all my other IP phones. 2. Ringer Volume. How do you set the ringer volume? So that it's set on reboot. Thanks ___ 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] (another) cisco 7960 question
> My 7960 is configured for two lines, and I can turn the other appearance > buttons into speed dials from the menus, but is there any way to program > the speed dials in the SIP.conf file? You can not: http://tinyurl.com/az4fp -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ 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] IAX losing registration
The problem is still occuring. it happens even if I register with myself, it works for some time and then just dies. The qualify still shows up as 65ms on the outside server, but the registry just says "Request Sent". and a reload doesn't help only restart. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Riddell Sent: Saturday, May 21, 2005 11:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX losing registration Joel Duffield wrote: > The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses "The > router uses NAT and TCP/IP port inspections" not stateful inspections. Make sure that your are using qualify=xxx for your IAX2 peers. For example, if you set it to 400 (this is in iax.conf in the definition for a particular account), it would send a request every 400ms (and mark the peer as unreachable if it goes over this amount). If you still get problems, lower the number. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 ___ 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 based Call Accounting software - 1st release
I would like to know, if I use your solution and I follow the instructions provided to get it to work with asterisk, what happens if: 1. The windows machine the CAM software runs on is down? 2. Can I use ForkCDR? will it update the CDR within CAM with a new record? (I think this last one depends if CAM uses Asterisks CDR functions to write to a new file? or did it rewrite the CDR?) Thanks and please reply. On 4/8/05, San Singhania <[EMAIL PROTECTED]> wrote: > > Hello Asterisk community, > > After numerous request from various companies where we have implemented * as > a phone system and also > from many other * users all over the world, yesterday we released the 1st > version of Asterisk module for > Call Accounting Mate (www.callaccounting.ws) . As some of you know we also > use Asterisk internally as our > phone system and as developers for Call Accounting Mate, we felt it was > necessary to implement a decent > Call Accounting software for *. Call Accounting Mate runs on Windows and is > completely web based. > It ships with the necessary source files and Asterisk modules to interface > Asterisk via tcpip to > Call Accounting Mate. > > We have set up a Asterisk - Call Accounting Mate forum so we can gather > input from the Asterisk > community. You can access the forum at > http://www.callaccounting.ws/forum/index.php?board=5.0 . > > Regards, > > San Singhania > www.callaccounting.ws > Tel : +1 718 5762066 > > ___ > 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] CallerID
Seems to me Im been displayed both... How can I control it? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 11:07 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |> -Original Message- |> I just tried call alert but something is wrong.. For each |call I get I |> see 2 or 3 events on the callerid.. The first is the actual number |> that dialed me, then 1 or 2 entries of my own number. |> |> Seems astapi or call alert is recognizing my own number is |if I called |> myself on each call. |> |> Is this an astapi issue, misconfiguration on my side or call alert |> problem? | |This can be a dialplan issue or a call-alert issue. It is |higly dependant on which channel you are monitoring and the |way callerid is handled throughout your dialplan. I have seen |similar things when using TAPIrex instead of Call-Alert. The |TAPI messaging layer allows you to peek into a lot of |environment things of the call, including the CALLERID and the |CALLEDID (subtle difference ;). | |Florian | | |___ |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] CallerID
Any tips on how to use astapi to popup something with callerid on a PC? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 11:05 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |> -Original Message- |> What do you mean "With ASTTAPI you can see events for your own phone |> too." | |As opposed to having something message you from the dialplan |you can make use of the manager events, that's the point I was |trying to make. | |> I already have astapi installed .. Have you tried call alert? |> Does it work |> as promised? | |Yes, it works as promised in my setup. | | |Florian | | |___ |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] LiveVOIP
[EMAIL PROTECTED] wrote: Is anyone having problems with LiveVOIP for outbound calls since their network upgrade a week ago? Ever since the network upgrade, it takes 2-3 times MINIMUM in order for a call to go from my system to theirs. I haven't changed any configs on my side, it just says " "call accepted by blah blah blah" and stays there for about 25 seconds, then comes back and says no one is available to answer at this time. When a call does go through, it gives back the message "call is making progress blah blah blah" As an interesting data point, my calls using VoipJet are doing the exact same thing, also newly as of about a week-10 days ago. It is very intermittent, though. I wonder if it's something happening at one of the interconnects that might possibly be shared by several ITSPs? B. ___ 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] G729 codec
At the cli prompt type “show codecs”. In the right hand column it states G.729A. Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of todd Sent: Sunday, May 22, 2005 12:55 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] G729 codec Hi all I have a question and hope it has not been answered before. I have searched the forums and mail but have not seen this answered conclusively. Does the G729 codec and licenses which digium sales for asterisk use g729 a or b or both; I have had a hard time getting a conclusive answer. If it does use g729b how could I show evidence to a client that it is b and not a? Thanks Sincerely TKG ___ 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
They talk the talk but don't deliver, some people have had good luck with them, I am not one of them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Sunday, May 22, 2005 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] LiveVOIP Our call quality is so bad we stopped using their service. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] LiveVOIP
Our call quality is so bad we stopped using their service. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] realtime excessive database queries
The queries you speak of are not native to realtime. They are part of pbx as a whole. And yes, the core programmers are aware of it and yes, they have admitted they don't like it. > Why not just load all the extensions into memory, and then have a > mechanism to mark an extension as stale and only then reload from the > database? Seems that with a basic caching mechanism you get all the > benefits of realtime without the downside of taking a performance hit > like it does now when having to go to the database so much. > > If I'm reading the code right the extensions are all in a linked list. > So it seems you could just add another item to the list to hold the > state of the extension, and when accessing the first item in the list > check the state. If it's stale and realtime is in use, update the > extension from the database and restart at the top of the list. > > Chris _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] (another) cisco 7960 question
My 7960 is configured for two lines, and I can turn the other appearance buttons into speed dials from the menus, but is there any way to program the speed dials in the SIP.conf file? ___ 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] G729 codec
Hi allI have a question and hope it has not been answered before. I have searched the forums and mail but have not seen this answered conclusively.Does the G729 codec and licenses which digium sales for asterisk use g729 aor b or both; I have had a hard time getting a conclusive answer. If it does use g729b how could I show evidence to a client that it is b and not a?ThanksSincerelyTKG ___ 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: Audio flutter on OH323 output?
In article <[EMAIL PROTECTED]>, I wrote: > > I've now done some RTP analysis of affected streams, and have found that > the times when we get the audio flutter correspond with parts of the > analysis showing repeated packets in the RTP stream. This only happens > on the outgoing stream; the incoming stream from the switch is perfect. > > When a packet is repeated, it is identical to the previous and occurs > about 7 to 14 microsec after, as if something in the software has decided > to send the packet twice in immediate succession. I've been trying to read through the openh323 and pwlib code related to writing the RTC stream. It's like a cross between peeling an onion and wading through treacle :-( I've focussed on the function RTP_UDP::WriteData() in openh323/src/rtp.cxx where dataSocket->WriteTo() is called in a loop. Is it possible that there are occasions where the packet did get sent but for some reason WriteTo() returned an error? I'm clutching at straws really, but without solving this, the system is not good enough to deploy. How can I turn on the PTRACE output from openh323, and where would it be sent to? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Newbie on IVR
Thanks, it sure did. - Original Message - From: "Julius Igugu" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, May 20, 2005 7:07 PM Subject: RE: [Asterisk-Users] Newbie on IVR > This should help. > > http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN654 > > > --- Jay Milk <[EMAIL PROTECTED]> wrote: > > Where's the question? > > > > -Original Message- > > From: Mike-Olumide, Johnson [mailto:[EMAIL PROTECTED] > > Sent: Friday, May 20, 2005 7:11 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Newbie on IVR > > > > > > Hi, > > > > I get fascinated when I dial someone and get an IVR play " for accounts > > department press 1, for sales, press 2 or hold the line for an operator" > > and then have MOH play before the line is picked up at the desired > > extesion. > > > > Please, permit me as I know this will be one of the dumbest questions to > > ask in a group like this. I'll apprecaite any specific > > guide/instruction. > > > > Thanks in anticipation. > > > > Mike > > > > > > Discover Yahoo! > > Get on-the-go sports scores, stock quotes, news & more. Check it out! > > > > ___ > > 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 > > > > Julius Igugu > SouthWork Co. Ltd. > > > > __ > Yahoo! Mail Mobile > Take Yahoo! Mail with you! Check email on your mobile phone. > http://mobile.yahoo.com/learn/mail > ___ > 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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk Project Consultant/Parner Wanted
Hello All, How are you all doing today? Good I hope. I am sure that I have asked this question before, but recently lost my emails server and thus any replies that you may have sent me. We are working to get a small online VoIP service established and I am looking for someone who might like to partner on this project or possibly offer reasonable consulting services. We need someone to take the lead on the development of the Asterisk PBX server and site configuration to get the service set up and operating. Please send an email if interested. Have a good day, Lonnie ___ 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] Fax and Voice VoIP services
Hello All, How are you all doing today? Good I hope. I am sure that I have asked this question before, but recently lost my emails server and thus any replies that you may have sent me. I am looking for an inexpensive provider that can offer: 1. one voice line with multiple voicemail boxes for Customer Service, Billing, Support, Abuse boxes. 2. one fax line to receive/send faxes which we can have delivered to our email address on our own server. Thanks in advance, Lonnie Cumberland ___ 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-oh323: Max simultaneous calls ?
Hi All, There is a parameter simultaneousMax=10 in oh323.conf. Had anybody tried out what is the maximum value that can be achieved ? What is the maximum number of simultaneous h323 calls can the oh323 driver can handle. I tried to get it only till 30 to 40 simultaneous calls. Anybody achieved better figures than this ? or have any idea how the oh323 can be tuned to get better values ? Thanks, ~Vamsi ___ 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
RES: [Asterisk-Users] Mysql CDR
I used the exemple listed in voip-info, but this list is wrong. I run de debug and I find de correct table, the error was in INSERT mysql command. Now all is function. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Matt Riddell Enviada em: domingo, 22 de maio de 2005 00:28 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Mysql CDR Rodrigo Otavio de Fraga wrote: > Hi, > When I finished a call, the asterisk give a message : FAILED TO INSERT > INTO DATABASE. Make sure that the details inside cdr_mysql.conf are correct. I.E. if it has username bob, password fred, host 127.0.0.1, run mysql -u bob -p Then it will ask you for a password. Type fred (or whatever your password is). Then it should connect (if the user/pass is in the db). If you then type use cdr (where cdr is the name of your cdr database) it should work. So, if you go through these steps, you should be able to find out where the problem is. (BTW: you did create the tables didn't you?) :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] CallerID
Hi, > -Original Message- > I just tried call alert but something is wrong.. For each > call I get I see 2 > or 3 events on the callerid.. The first is the actual number > that dialed me, > then 1 or 2 entries of my own number. > > Seems astapi or call alert is recognizing my own number is if I called > myself on each call. > > Is this an astapi issue, misconfiguration on my side or call > alert problem? This can be a dialplan issue or a call-alert issue. It is higly dependant on which channel you are monitoring and the way callerid is handled throughout your dialplan. I have seen similar things when using TAPIrex instead of Call-Alert. The TAPI messaging layer allows you to peek into a lot of environment things of the call, including the CALLERID and the CALLEDID (subtle difference ;). Florian ___ 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] CallerID
Hi, > -Original Message- > What do you mean "With ASTTAPI you can see events for your > own phone too." As opposed to having something message you from the dialplan you can make use of the manager events, that's the point I was trying to make. > I already have astapi installed .. Have you tried call alert? > Does it work > as promised? Yes, it works as promised in my setup. Florian ___ 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] CallerID
I just tried call alert but something is wrong.. For each call I get I see 2 or 3 events on the callerid.. The first is the actual number that dialed me, then 1 or 2 entries of my own number. Seems astapi or call alert is recognizing my own number is if I called myself on each call. Is this an astapi issue, misconfiguration on my side or call alert problem? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 06:04 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |> -Original Message- |> Anton Krall wrote: |> > What re you guys doing for windows callerid from Asterisk |> besides using yac? |> > |> > Any other working software? | |With ASTTAPI you can see events for your own phone too. |http://sourceforge.net/projects/asttapi/ | |Take a look at this client: |http://www.ivrsoft.com/call-alert.htm | | | |___ |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
Is anyone having problems with LiveVOIP for outbound calls since their network upgrade a week ago? Ever since the network upgrade, it takes 2-3 times MINIMUM in order for a call to go from my system to theirs. I haven't changed any configs on my side, it just says " "call accepted by blah blah blah" and stays there for about 25 seconds, then comes back and says no one is available to answer at this time. When a call does go through, it gives back the message "call is making progress blah blah blah" ___ 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] CallerID
What do you mean "With ASTTAPI you can see events for your own phone too." ?? I already have astapi installed .. Have you tried call alert? Does it work as promised? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 06:04 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |> -Original Message- |> Anton Krall wrote: |> > What re you guys doing for windows callerid from Asterisk |> besides using yac? |> > |> > Any other working software? | |With ASTTAPI you can see events for your own phone too. |http://sourceforge.net/projects/asttapi/ | |Take a look at this client: |http://www.ivrsoft.com/call-alert.htm | | | |___ |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] CallerID
Very interesting approach Matt, I coincidently have centericq installed on
my asterisk server so I just tried that configuration and worked very nice!
Do you know of any windows apps that might be able to do this? As you noted,
winxp doesnt have the popup app and sometimes msn or icq takes too long to
reach its destination.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matt Riddell
|Sent: Sábado, 21 de Mayo de 2005 11:01 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] CallerID
|
|Anton Krall wrote:
|> What re you guys doing for windows callerid from Asterisk
|besides using yac?
|>
|> Any other working software?
|
|I use:
|
|MSN Messenger (this is a bit slow - uses centericq)
|===
|
|exten => s,2,System(/bin/echo -e 'Incoming Call From: ${CALLERIDNAME},
|${CALLERIDNUM} To: ${ARG3} Received:
|${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at ${DATETIME:9}' |
|centericq -s msg -p msn -t [EMAIL PROTECTED])
|
|Windows Popup (fast but missing on XP - uses smbclient)
|===
|
|exten => s,1,System(/bin/echo -e 'Incoming Call From:
|${CALLERIDNAME},
|${CALLERIDNUM} To:${ARG3} Received:
|${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at
|${DATETIME:9}'|/usr/bin/smbclient -M ${ARG4})
|
|SMS to my cellphone (only if I missed a call - uses smsx AGI)
|=
|
|exten => s,10,AGI(smsx|64211387245|txt|You missed an incoming
|call from
|${CALLERIDNAME} - ${CALLERIDNUM} to ${ARG3} on
|${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at ${DATETIME:9})
|
|Sorry about the line splits :)
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html)
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
|___
|Asterisk-Users mailing list
|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asterisk-users
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| http://lists.digium.com/mailman/listinfo/asterisk-users
|
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Re: [Asterisk-Users] Getting a Cisco gateway to work with Asterisk
When you say identify I presume you are trying to get the Cisco to register as a user. To the best of my knowledge it cannot do this. Instead define a peer in sip.conf which is the gateway and place traffic matching this peer into a context that is defined in your extensions.conf file. The Cisco will need dial-peer statements to match inbound dialed digits and forward all matching calls to your Asterisk box. Mark Dutton wrote: Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] The gateway works with other SIP servers that don't require authentication, but Asterisk wants it to authenticate, or at least idenitify itself and I cannot work this bit out. If I put in the host address in my sip.conf, I still get a "cannot find host 192.168.44.23:, where number> is actually some random port number. I am at my wits end. Regards Mark ___ 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 to pass Asterisk -SIP- Cisco AS -H323- world ?
Hello, i know thats is * mailing list but maybe here are cisco guru's which can help. My network schema is: Softphones <-SIP-> Asterisk <-SIP->\/<-H323-> WORLD2 > Cisco AS5350 <- E1 -> WORLD (Phones <-> Traditional PBX <-E1->--/) - will be developed soon. Communications beetween WORLD and softphones works well but i have an H323 link to other site and i want to allow calling from softthones (in future from Phones too) calling to the WORLD2. I tried to add dialpeers but this doesnt work - all calls are routed via E1 to WORLD. This is part of my config: ! GK Config: interface FastEthernet0/0 ip address 192.168.X.X 255.255.255.0 duplex auto speed auto h323-gateway voip interface h323-gateway voip id TGK1 ipaddr 194.X.X.X 1719 h323-gateway voip h323-id MYID ! gateway ! For World -> Softphones communication dial-peer voice 14 pots incoming called-number 2323. direct-inward-dial ! dial-peer voice 15 voip destination-pattern 2323. session protocol sipv2 session target ipv4:192.168.X.X codec g711alaw ! For outgoing Softphones - World dial-peer voice 1000 pots application session destination-pattern .T direct-inward-dial port 3/1:D forward-digits all ! i tried to add ! dial-peer voice 999 voip application session destination-pattern 0. target session ras ! but all calls are still routed via dial peer 1000 - why ? I want to pass all calls thru cisco becouse i need one point for billing for asterisk and PBX calls and in future i need to make calls from PBX to the WORLD2 destinantion. PLEASE HELP! Thanks, Adam ___ 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: Audio flutter on OH323 output?
In article <[EMAIL PROTECTED]>, I wrote: > In article <[EMAIL PROTECTED]>, > Michael Manousos <[EMAIL PROTECTED]> wrote: > > > > Can you get an ethereal trace on a call with that problem? > > Run an RTP analysis on the captured stream (Tools Menu) and save > > the contents of the RTP packets in audio files. Then check if > > the playback of these files is normal or not. > > Aha, sounds like Ethereal has even more clever features - I didn't > know it could do that. > > Fortunately, I've been running a continuous tcpdump capture on the > Asterisk box of the traffic between it and the switch. I'll see > what Ethereal thinks of it. Hi Michael, I've now done some RTP analysis of affected streams, and have found that the times when we get the audio flutter correspond with parts of the analysis showing repeated packets in the RTP stream. This only happens on the outgoing stream; the incoming stream from the switch is perfect. When a packet is repeated, it is identical to the previous and occurs about 7 to 14 microsec after, as if something in the software has decided to send the packet twice in immediate succession. I've pasted below an extract from the CVS file saved from the RTP stream analysis. I also saved the payload (pity it converts it to uLaw instead of saving it in the stream's native format) and listened to it to confirm the periods of distortion. I would assume the problem is somewhere in the depths of openh323, but any pointers in the right direction would be appreciated! Cheers Tony Here is the CSV extract: Packet,Sequence,Delay (s),Jitter (s),Marker,Status,Date,Length 144209,17353,0.02,0.03,,[ Ok ],05/17/2005 17:05:48.420,214 144225,17354,0.020006,0.03,,[ Ok ],05/17/2005 17:05:48.440,214 144226,17354,0.07,0.03,,Wrong sequence nr.,05/17/2005 17:05:48.440,214 144244,17355,0.019987,0.04,,[ Ok ],05/17/2005 17:05:48.460,214 144260,17356,0.019998,0.04,,[ Ok ],05/17/2005 17:05:48.480,214 144261,17356,0.08,0.04,,Wrong sequence nr.,05/17/2005 17:05:48.480,214 144276,17357,0.019991,0.04,,[ Ok ],05/17/2005 17:05:48.500,214 144296,17358,0.019992,0.04,,[ Ok ],05/17/2005 17:05:48.520,214 144298,17358,0.09,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.520,214 144313,17359,0.019990,0.05,,[ Ok ],05/17/2005 17:05:48.540,214 144329,17360,0.02,0.05,,[ Ok ],05/17/2005 17:05:48.560,214 144349,17361,0.019992,0.05,,[ Ok ],05/17/2005 17:05:48.580,214 144368,17362,0.020003,0.05,,[ Ok ],05/17/2005 17:05:48.600,214 144369,17362,0.06,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.600,214 144383,17363,0.019990,0.05,,[ Ok ],05/17/2005 17:05:48.620,214 144401,17364,0.019995,0.05,,[ Ok ],05/17/2005 17:05:48.640,214 144417,17365,0.020001,0.05,,[ Ok ],05/17/2005 17:05:48.660,214 144432,17366,0.019994,0.05,,[ Ok ],05/17/2005 17:05:48.680,214 144452,17367,0.019997,0.05,,[ Ok ],05/17/2005 17:05:48.700,214 144469,17368,0.020002,0.05,,[ Ok ],05/17/2005 17:05:48.720,214 144486,17369,0.019998,0.05,,[ Ok ],05/17/2005 17:05:48.740,214 144487,17369,0.08,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.740,214 144506,17370,0.019985,0.05,,[ Ok ],05/17/2005 17:05:48.760,214 144523,17371,0.01,0.05,,[ Ok ],05/17/2005 17:05:48.780,214 144540,17372,0.020005,0.05,,[ Ok ],05/17/2005 17:05:48.800,214 144559,17373,0.019992,0.05,,[ Ok ],05/17/2005 17:05:48.820,214 144560,17373,0.08,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.820,214 144576,17374,0.019988,0.06,,[ Ok ],05/17/2005 17:05:48.840,214 144577,17374,0.07,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.840,214 144592,17375,0.019992,0.06,,[ Ok ],05/17/2005 17:05:48.860,214 144593,17375,0.08,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.860,214 144610,17376,0.019987,0.07,,[ Ok ],05/17/2005 17:05:48.880,214 144627,17377,0.019997,0.06,,[ Ok ],05/17/2005 17:05:48.900,214 144628,17377,0.08,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.900,214 144642,17378,0.019989,0.07,,[ Ok ],05/17/2005 17:05:48.920,214 144661,17379,0.019997,0.07,,[ Ok ],05/17/2005 17:05:48.940,214 144678,17380,0.020004,0.06,,[ Ok ],05/17/2005 17:05:48.960,214 144679,17380,0.09,0.07,,Wrong sequence nr.,05/17/2005 17:05:48.960,214 144695,17381,0.019988,0.07,,[ Ok ],05/17/2005 17:05:48.980,214 144696,17381,0.07,0.07,,Wrong sequence nr.,05/17/2005 17:05:48.980,214 144713,17382,0.019982,0.08,,[ Ok ],05/17/2005 17:05:49.000,214 144732,17383,0.020002,0.07,,[ Ok ],05/17/2005 17:05:49.020,214 144733,17383,0.08,0.07,,Wrong sequence nr.,05/17/2005 17:05:49.020,214 144747,17384,0.019989,0.08,,[ Ok ],05/17/2005 17:05:49.040,214 144768,17385,0.02,0.07,,[ Ok ],05/17/2005 17:05:49.060,214 144769,17385,0.07,0.07,,Wrong sequence nr.,05/17/2005 17:05:49.060,214 144784,17386,0.019986,0.07,,[ Ok ],05/17/2005 17:05:49.080,214 144801,17387,0.020001,0.07,,[ Ok ],05/17/2005 17:05:49.100,214 144802,17387,0
Re: [Asterisk-Users] Boosting Internet Bandwidth for VOIP
chawki hammoud wrote: --- Doug Lytle <[EMAIL PROTECTED]> wrote: ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists My knowledge about this topic is limited. I am not aware about any QoS running. Do you know how I can find out? These are the errors I get if I try to run my Wonder Shaper script twice on the same interface. For me that would be tun0. The script shows the following to remove the current TC queues: tc qdisc del dev $DEV root tc qdisc del dev $DEV ingress Make sure $DEV is your actual network interface that you were getting the errors, before trying the new script. For example: tc qdisc del dev eth0 root tc qdisc del dev eth0 ingress This will probably get rid of at least the 'File exists' errors, not sure on the others. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 Asterisk boxes sharing dial plans.
Hello All, I have two asterisk boxes. 1 for home and 1 for work/ham radio club. I have 2 SIP trunks on each server. What is the best way to share the trunks? Thanks, David PS FWDOUT is great!!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pri doesn't accept Zap/g2 to call
I have a Sangoma Card with two PRIs. They are both
configured in Zaptel and Zapata;
In Zapata I have them separated in Group 1 and 2 but if I
make a call and specify Zap/g2 it doesn’t go when calling Channels :
HERE IS what I get:
Accepting AUTHENTICATED call from x.x.x.x
> requested format = speex,
> requested prefs =
(),
> actual format =
gsm,
> host prefs =
(ilbc|gsm),
> priority = mine
-- Executing
Dial("IAX2/[EMAIL PROTECTED]", "Zap/g2/3337885836|100|T")
in new stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called g2/3337885836
-- Channel 0/1, span 2 got hangup request
-- Hungup 'Zap/32-1'
== No one is available to answer at this time
(1:0/0/0)
-- Executing Hangup("IAX2/[EMAIL PROTECTED]",
"") in new stack
== Spawn extension (default, 3337885836, 2) exited
non-zero on 'IAX2/[EMAIL PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
Zapata.conf
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
[default]
exten => _X.,1,Dial(${PRITRUNK2}/${EXTEN},100,T)
exten => _X.,2,Hangup
[firefly1]
exten => _X.,1,Dial(${PRITRUNK1}/${EXTEN},100,T)
exten => _X.,2,Hangup
Zaptel.conf:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-61
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[Asterisk-Users] Questions about TE410P card
Hello i have two questions 1 - is there a maximum number of TE410P/P.C. (don't take the number of pc's pci slots in your account)? 2- if i installed two TE410P cards in the server, what should the server internet connection bandwidth be? . Regards Yousri ___ 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] Getting a Cisco gateway to work with Asterisk
Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:[EMAIL PROTECTED] I cannot figure out how to get it to identify as sip:[EMAIL PROTECTED] The gateway works with other SIP servers that don't require authentication, but Asterisk wants it to authenticate, or at least idenitify itself and I cannot work this bit out. If I put in the host address in my sip.conf, I still get a "cannot find host 192.168.44.23:, where is actually some random port number. I am at my wits end. Regards Mark ___ 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] *@home 1.0 FWD inbound problems, 2 calls generated
Hi ALL
Have installed [EMAIL PROTECTED] 1.0
On FWD DID's, appears that 2 calls are generated to the inbound extention. I
have confirmed this on a number of friends boxes also. Does anyone have a fix
for this ? I set the DID simply to a custom context and it did the same...
Anyone have a way to fix this ?
Here is the output..
-- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4,
actual format = 4
-- Executing Goto("IAX2/[EMAIL PROTECTED]/5", "ext-local|7020|1") in new
stack
-- Goto (ext-local,7020,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/5", "exten-vm|[EMAIL
PROTECTED]|7020") in
new stack
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/5", "FROMCONTEXT=exten-vm") in
new
stack
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/5", "0?novm|1:3") in new stack
-- Goto (macro-exten-vm,s,3)
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/5", "0?novm|1") in new stack
-- Executing Macro("IAX2/[EMAIL PROTECTED]/5", "dial|15|tr|7020") in new
stack
-- Executing AGI("IAX2/[EMAIL PROTECTED]/5", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4,
actual format = 4
-- Executing Goto("IAX2/[EMAIL PROTECTED]/6", "ext-local|7020|1") in new
stack
-- Goto (ext-local,7020,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/6", "exten-vm|[EMAIL
PROTECTED]|7020") in
new stack
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/6", "FROMCONTEXT=exten-vm") in
new
stack
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/6", "0?novm|1:3") in new stack
-- Goto (macro-exten-vm,s,3)
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/6", "0?novm|1") in new stack
-- Executing Macro("IAX2/[EMAIL PROTECTED]/6", "dial|15|tr|7020") in new
stack
-- Executing AGI("IAX2/[EMAIL PROTECTED]/6", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 1
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1116763505.28
-- dialparties.agi: channel = IAX2/[EMAIL PROTECTED]/5
-- dialparties.agi: callerid = 0409839735
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = IAX2
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID name and number are '0409839735'
-- dialparties.agi: Added extension 7020 to extension map
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: Extension 7020 cf is disabled
-- dialparties.agi: Extension 7020 do not disturb is disabled
-- dialparties.agi: priority = 1
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1116763505.29
-- dialparties.agi: channel = IAX2/[EMAIL PROTECTED]/6
-- dialparties.agi: callerid = 0409839735
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = IAX2
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID name and number are '0409839735'
-- dialparties.agi: Added extension 7020 to extension map
-- dialparties.agi: Extension 7020 cf is disabled
-- dialparties.agi: Extension 7020 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 7020 has call waiting disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 7020 has call waiting disabled
-- dialparties.agi: DbSet CALLTRACE/7020 to 0409839735
dialparties.agi: Dial string is SIP/7020|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("IAX2/[EMAIL PROTECTED]/6", "SIP/7020|15|tr") in new stack
-- Called 7020
-- dialparties.agi: DbSet CALLTRACE/7020 to 0409839735
dialparties.agi: Dial string is SIP/7020|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("IAX2/[EMAIL PROTECTED]/5", "SIP/7020|15|tr") in new stack
-- Called 7020
-- SIP/7020-22d9 is ringing
-- SIP/7020-4abd is ringing
-- Nobody picked up in 15000 ms
-- Executing Wait("IAX2/[EMAIL PROTECTED]/6", "1") in new stack
-- Nobody picked up in 15000 ms
-- Executing Wait("IAX2/[EMAIL PROTECTED]/5", "1") in new stack
== Spawn extension (macro-exten-vm, s, 5) exited non-zero on
RE: [Asterisk-Users] CallerID
Hi, > -Original Message- > Anton Krall wrote: > > What re you guys doing for windows callerid from Asterisk > besides using yac? > > > > Any other working software? With ASTTAPI you can see events for your own phone too. http://sourceforge.net/projects/asttapi/ Take a look at this client: http://www.ivrsoft.com/call-alert.htm ___ 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] Upgrade cause's no Audio on IAX
Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction. Does anyone know what would have changed to cause this or what I would need to do to look at solving the issue ? I am now offline :( and for some reason rolling back to the older version now does not want to run :( My IAX conf [general] tos=lowdelay jitterbuffer=no disallow=all allow=speex allow=ilbc allow=gsm register => user:[EMAIL PROTECTED] [guest] type=user context=default auth=none [2347] type=friend username=user secret=password auth=md5 host=gw1.austechpartnerships.com context=default trunk=yes qualify=3000 disallow=all allow=ilbc Thanks 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
Re: [Asterisk-Users] Working Xten, Asterisk, double-NAT configs out there?
> I have my * box NAT'd with all ports forwarded that are SIP related > (based on Wiki). I also have nat=yes, externalip=WAN address of > firewall, internalip=LAN network of *. > > I have my Xten soft phone on a PC which is NAT'd behind firewall with > ports forwarded. I have also followed instructions on Wiki for Xten. Take a look here: http://willypick.mindsay.com/?entry=10 Your problem does not sound like NAT to me, but authentication on the other end. Max retries refers to the phone you are trying to reach. ___ 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] Error running Make config on Debian Sarge
On Mon, May 16, 2005 at 12:53:49PM -0500, Ben Johnson wrote: > I am running Asterisk 1.0.7 on Debian Sarge RC3. When I attempt to run > "make config" to create the zaptel boottime script I receive the following Frankly, you don't need to. Chances are /etc/rcS.d/S40hotplug will detect your card's module and modprobe it. modprobe will then run ztcfg as per the instruction in /etc/modules.conf /etc/modprobe.conf . If this is not the case, then add themodule to /etc/modules, and the module will be modprobed nevertheless. Reasons why the module won't be discovered: * The module is ztdummy (you don't have a card) * You disabled hotplug (Not a very bad move) * Hotplug did not discover your card (this is probably a bug, and worth reporting) > > if [ -d /etc/rc.d/init.d ]; then \ >install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ >chkconfig --add zaptel; \ > elif [ -d /etc/init.d ]; then \ >install -D -m 755 zaptel.init /etc/init.d/zaptel; \ >chkconfig --add zaptel; \ > fi totally irrelevant to debian. However there is a part in the makefile that should have replaced 'chkconfig' with ':', right. I wonder is there is any warning above about redefinition of "CHKCONFIG" (my private crusade, sorry) Actually in my rapid package I disabled running ztcfg at modprobe time and did add an init.d script that simply runs ztcfg. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Help compiling zaptel in Debian
On Sun, May 15, 2005 at 08:15:10PM -0600, Andres Paglayan wrote: > follow this link > ignore the German and see the commands > http://www.vonloesch.de/node/17 > > for the last part be sure that you modprobe the right driver for your > particular device. I don't read german, but those instructions are slightly out-of-date. Specifaically, the current kernel varion in Sarge is 2.4.27-*2* and not *1* . The debian packages also try to provide you nicer to build zaptel packages. (And did I ever mention I have them pre-built at http://tzafrir.org.il/rapid/ ?) > > one little thing is that in Debian you shouldn't use /usr/local/bin, but > /usr/bin, if you are using the source from digium you might to search > and replace within Makefiles Actually, if you build binaries on your own and not from packages, they should generally go to under /usr/local per the Debian policy. That way it is guaranteed to never colide with any later package, for instance. The same holds for most other distros. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] realtime excessive database queries
Does asterisk really need to be doing 3 database calls for every priority in every extension? Why not just load all the extensions into memory, and then have a mechanism to mark an extension as stale and only then reload from the database? Seems that with a basic caching mechanism you get all the benefits of realtime without the downside of taking a performance hit like it does now when having to go to the database so much. If I'm reading the code right the extensions are all in a linked list. So it seems you could just add another item to the list to hold the state of the extension, and when accessing the first item in the list check the state. If it's stale and realtime is in use, update the extension from the database and restart at the top of the list. 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] Boosting Shared Internet Bandwidth for Asterisk
Depending on you bandwidth, you might not need QoS. Priority could be enough. In you sip.conf (if you use SIP), place a tos value: [general] tos = 0x10 ; low delay or tos = 0x46 ; DiffServ Premium (EF: Expedited Forward) Remark: for un unknown reason, tos=lowdelay doesn't work anymore on my asterisk (v1.0.7), but was working in the past. I replaced it by 0x10 (hex value of lowdelay). Most of the routers support PFIFO (FIFO with priority), which means that low delay flagged packet will be sent in priority. I haven't tested the 0x46 value yet. Routers must be configured for DiffServ values, while ToS is by default. But the low delay TOS bit is also set within the 0x46 value. If a router treat the the DiffServ byte as TOS, it should be sent with priority as well (to be validated). If you want to check what priority is set inside your packets, you might use Ethereal. You might see either UDP or RTP packets, depending on the RTP ports that are used. In the branch "Internet Protocol", you will find the TOS/DiffServ decode, named "Type of service" or "Differential services Field". The TOS low delay bit is the 5th, and should be 1. If you have a low bandwidth connection (e.g. 600/100), you might have a new problem if you are using TOS as low delay. Voice will be good, but data will stall. QoS won't resolve it, because big packets take too much time to travell. The only way to share bandwidth for voice and data, on low bandwidth lines, is to fragment the data. An MTU of 700 is quite good, but you have to assume about 15% of bandwidth loss, because of twice more overheads on big packets. Allthough, a 1200/200 kbps line usually doesn't require such tricks. Remark about Grandstream: If you are using a GS device, you must know that QoS is buggy, and will have no effect at all. You must upgrade to the beta version of the firmware, which is OK. Therefore, GS recommands a QoS value of 48 (whithout "0x" on a GS device). This is a DiffServ value, which does not set the los delay TOS bit. Cisco recommands 46, which does. Jean-Christophe chawki hammoud a écrit : >--- Matt Riddell <[EMAIL PROTECTED]> wrote: > > > > >>Assuming your provider completely ignores QOS, it is >>still not a >>complete waste of time. >> >>If for example you have 5 people on the LAN, 4 >>uploading files to a >>remote server and 1 trying to make a phone call. >> >> > >My ISP has the internet connection set-up where 8 >people share the bandwidth. Would the script still >help boost my voip calls? > > > > > >__ >Yahoo! Mail Mobile >Take Yahoo! Mail with you! Check email on your mobile phone. >http://mobile.yahoo.com/learn/mail >___ >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] Boosting Internet Bandwidth for VOIP
--- Doug Lytle <[EMAIL PROTECTED]> wrote: > >./rc.tc start > >RTNETLINK answers: File exists > >RTNETLINK answers: File exists > > > > > Looks like you are already running some type of QoS > script, you'll need > to stop it did befor trying the new script. > > Doug My knowledge about this topic is limited. I am not aware about any QoS running. Do you know how I can find out? Chawki > > ___ > 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 > __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ 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
