[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 available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register asterisk1 on asterisk2 and asterisk2 on asterisk1 ? Thanks Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 musiconhold loaded. Is that the correct way to disable it Is there a way to get rid of this warning Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 but the digits are not captured. I tested with get data and it works I tested with exec background and it works The calls from the exterior use RFC2833. If I use inband, none of the 3 work. Why would stream file behave like this Here is the setup Pri SIP Gateway(Patton) - Asterisk (ztdummy) - SIP Phone Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 scripts. Patrick === 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 but the digits are not captured. I tested with get data and it works I tested with exec background and it works The calls from the exterior use RFC2833. If I use inband, none of the 3 work. Why would stream file behave like this Here is the setup Pri SIP Gateway(Patton) - Asterisk (ztdummy) - SIP Phone Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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://lists.digium.com/mailman/listinfo/asterisk-users
[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 connected to the 911 agent and be able to talk to him. The call is dropped only when the 911 agent hangs up on his side. Is there a way in asterisk to disable the hangup from the client after he has dialed 911 ? Or maybe asterisk can keep the channel up and call back the user to re-establish the call until the hangup comes from the other side Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 the call is semi-private ? Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 that Asterisk replace the name and number to unknown and then send the call to my Patton box. How can I make this setup work ? Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.digium.com/mailman/listinfo/asterisk-users
[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 with both units except for the username of course, Anybody ever worked with these units ? Our CMTS is a uBR7246VXR from Cisco Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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} /dev/${TTY}
Any idea what is the problem ?
here is a show channels before the crash
SIP/131-f5ad (None) Ringing AppDial((Outgoing
Line))
SIP/123-8bc1 [EMAIL PROTECTED]:1 RingDial(SIP/131|16|tr)
Zap/11-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/10-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/3-1 [EMAIL PROTECTED]:1 Up Bridged
Call(SIP/137-adb6)
SIP/137-adb6 [EMAIL PROTECTED]:2
Up Dial(Zap/G1/915143334233)
Zap/8-1 [EMAIL PROTECTED]:2 Up Queue(support|t|||300)
Zap/7-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
SIP/141-178a [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
SIP/141-f5f6 [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
SIP/141-5371 [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
Zap/4-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Agent/111[EMAIL PROTECTED]:1 Down(None)
Local/[EMAIL PROTECTED]
[EMAIL PROTECTED]:1Ring(None)
Local/[EMAIL PROTECTED]
[EMAIL PROTECTED]:1 Down(None)
Zap/6-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/2-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
The queue application seemed to be crashed because I got no output from a
show agents or a show queues
Patrick
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[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 answer, the calls go to the agent's voicemail. in the show queues I see Agent/108 (Not in use) I did the show queues while talking to the agent in question. Is this normal behaviour ? Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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_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_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_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 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position I had to disable logging to be able to use the console Anybody seen this one ? Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 solutions I read was to use the # to transfer the calls. We are already using the # and it still doesn't free up the agent. Any idea where to look next ? Is there a debug I could activate to see where is the problem ? Could this be context related ? Patrick ___ --Bandwidth and Colocation sponsored by Easynews.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] 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 appreciated. Patrick ___ --Bandwidth and Colocation sponsored by Easynews.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] 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 modules there is no noise if we put three fxo modules on the lines 1-2-3, we have noise on line 1 (Zap/1). line 2 and 3 have no noise if we put three fxo modules on the lines 2-3-4 we have no noise if we put 4 fxo modules we have noise on the line 1 if we use an older TDM card, revision E/F, there is no noise problem. Digium has no explanation for now and have asked for a RMA of one of the cards. I will keep you informed. If someone else has seen this behaviour, tell me if there is an explanation. Patrick ___ --Bandwidth and Colocation sponsored by Easynews.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] 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 And the static noise is gone !! Anobody have an idea why this happened Of course it doesn't solve my problem because we have one old card and several new card but it may give digium an idea of where is my problem Patrick 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 other items The only 2 items that we have not changed is the mobo and the power supply. At first it was on scsi drives but we re-installed using a IDE drive We deactivated the two onboard nic and tried two different brand. We have deactivated hyper-treading We have deactivated USB We have deactivated SATA We have tried a noise-cancelling power-bar We have tried two different phones lines We have tried several IP phones, Cisco, Snom, Gnet (There is no noise for a call between two phones) The phone is connected directly in the nic card so there is no network problem possible. We have tried several TDM Card Anybody knows if the motherboard or the power-supply could be the problem ? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 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 other items The only 2 items that we have not changed is the mobo and the power supply. At first it was on scsi drives but we re-installed using a IDE drive We deactivated the two onboard nic and tried two different brand. We have deactivated hyper-treading We have deactivated USB We have deactivated SATA We have tried a noise-cancelling power-bar We have tried two different phones lines We have tried several IP phones, Cisco, Snom, Gnet (There is no noise for a call between two phones) The phone is connected directly in the nic card so there is no network problem possible. We have tried several TDM Card Anybody knows if the motherboard or the power-supply could be the problem ? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 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 built in hands free operation display size codecs communication protocol (SIP, h.323) price availability reliability know bugs / limitations asterisk compatibility If someone has done this recently that would save me some time Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.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] 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-Users@lists.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] 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/remove PRI cards without shutting down the asterisk system Is there a solution that exist ? Someone told me to look at the C-PCI technology, it seems that telecom companies use this. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.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 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/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible hot-swappable PRI card and board ? Someone told me to look at the C-PCI technology, it seems that telecom company use this. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
