[asterisk-users] g729 setup help

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not

[asterisk-users] setcallerpres not working

Hi I have the following setup phone - mta - asterisk - patton_sn2400 - PRI I am trying to program *67 to block caller id name and number To do this correctly I have to leave the fields callerid name and number unchanged and only set the flag callerpres to restricted The problem seems to be

[asterisk-users] PRI to SIP

Hi Can someone recommend a PRI to SIP Box that work well with asterisk We are presently testing with a Patton Smartnode 2400 but we are unable to fax through it. We don't want to use digium card in a linux box for the PRI connection. Which Cisco box would work. Thanks Patrick

[asterisk-users] semi-private call

Hi Do you know if the SIP protocol is compatible with semi-private calls. I can contruct a private call by putting the SIP Privacy header to "id" and then sending the call to my SIP-Pri box and it works This tell my Pri provider that the call is private. How can I tell my Pri provider that th

[asterisk-users] disable client side hangup after dialing 911

Hi Has anyone tried to reproduce the following behavior that a standard phone line does with 911. Normally if someone calls 911 and hangs up after the call has been established then the line is not dropped because it is held by the 911 agent. If you pickup your phone you should still be con

[asterisk-users] echo cancellation and ztdummy

Hi Are echo cancellation parameters useful when using the ztdummy driver and no physical card ? Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://

[asterisk-users] stream file not working but get data and exec background work

Hi I have a strange problem. I use the agi command stream file for my vertical services like *98 If the call comes from a sip phone with dtmfmode=inband in sip.conf then it works. But if I call the same script from an external line the stream file doesn't work properly The audio is played

[asterisk-users] - SOLVED - stream file not working but get data and exec background work

Hi While testing I found a solution to my problem. I don't understand it maybe someone here can explain it. In my script, if I call a Playback just before my stream file then everything works ok. Without the playback then the digits are not captured I will playback a silence to patch my scri

[asterisk-users] disable musiconhold

Hi I would like to disable correctly musiconhold for my users when they are using the callwaiting feature. I have set in modules.conf noload => res_musiconhold.so Now I don't have music on hold when I use call waiting but I have this warning: -- Music class default requested but no musiconh

[Asterisk-Users] asterisk compatible, hot swappable PRI card

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remo

[Asterisk-Users] add/remove PRI card without rebooting

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remo

[Asterisk-Users] segmentation fault

Hi Asterisk died this morning with this message safe_asterisk: line 83: 6828 Segmentation fault (core dumped) asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY} Any idea what is the problem ? here is a show channels before the crash SIP/131-f5ad (None) Ringing AppDial

[Asterisk-Users] arris e-mta

Hi This may be off topic because it involve cable. I am testing with Arris cable modem / MTA I have 2 models, one older and one newer. With older one, everything works fine With the new one, I can register, make a call and I hear the other person but he can't hear me The config is the same

[Asterisk-Users] commercial package for vertical services

Hi Are there any packages to implement vertical services in asterisk commercial (or free) Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.d

[Asterisk-Users] phone comparison matrix

Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call buil

[Asterisk-Users] Snom phones - any advice

Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Us

[Asterisk-Users] static noise with this hardware any advice

Hi We have static noise problem on our asterisk server. latest stable release. The card is a new TDM04B We have it installed on the following hardware Motherboard Intel SE7520BD2SCSI 2x POWER SUPPLY 730W INTEL I will not mention the other hardware because we have desactivated/changed all the

[Asterisk-Users] static noise with this hardware any advice

Follow-up on this We have tried several things without success. Digium responded that the problem was NMI (non-maskable interrupts) and told me to boot linux with the nmi_watchdog=0 option It did not solve the problem. Finally I replaced the TDM card with an older one (revision F) 2FXO 2FXS

[Asterisk-Users] what is your echo solution

Hi Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. We have asterisk 1.2.0 3 x digium TDM400P 30 Snom320 + 5 Snom360 For now the best setup I have is using Mark2 Echo cancel. Thanks Patric

[Asterisk-Users] warning message

Hi I got this warning message repeating itself in the log this morning Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_

[Asterisk-Users] calls forwarded to busy agent

Hi We have a call queue setup with several agents using agentcallbacklogin. If one of the agent is logged in and is talking on the phone with another employee the queue application doesn't see that the phone is busy and continues to forward incoming calls to him. Since the agent cannot answe

[Asterisk-Users] hardware echo cancellation for TDM card

Hi Just checking, Is there any hardware echo cancellation card available for the digium TDM400P card I read the archives and could not find any. I think I need the TDM2400 card for this Thanks Patrick ___ --Bandwidth and Colocation provided by E

[Asterisk-Users] how to follow a call in the console

Hi Is there a way to easily follow a call in the console log Maybe by adding a unique call ID or something Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] static noise - follow up

Hi two weeks ago I posted a message concerning static noise on our asterisk system we have made a bunch of tests and these are the results We use a TDM card revision I and on the card there is a sticker that says revision G If we put one fxo modules there is no noise if we put two fxo module

[Asterisk-Users] voicetronix openline4 comments

Hi I would like your comments on the openline4 card from voicetronix. I am trying to get one working and find it difficult. I was able to get asterisk working yesterday but now it doesn't work anymore While it worked I was able to make some calls and I heard a lot of jitter Any comments appre

[Asterisk-Users] agent transfer problem

Hi We have this problem: We have a queue with several agents logged using agentcallbacklogin If an agent receives a call and then transfer it to another agent or to another employee or to another queue, the call remains connected to the original agent. I read the archives and all the soluti